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integrated new version of CAPS and fixed miscompilation with GCC 4.3

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git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/branches/lmms-mv@726 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
Tobias Doerffel
2008-02-27 17:52:56 +00:00
parent a7b14c6660
commit f83aecf617
60 changed files with 21649 additions and 869 deletions
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253
plugins/ladspa_effect/caps/Amp.cc
View File

@@ -1,11 +1,14 @@
/*
Amp.cc
Copyright 2003-6 Tim Goetze <tim@quitte.de>
Copyright 2003-7
Tim Goetze <tim@quitte.de>
David Yeh <dtyeh@ccrma.stanford.edu> (Tone Stack in TS models)
http://quitte.de/dsp/
tube amplifier models
Tube amplifier models
*/
/*
This program is free software; you can redistribute it and/or
@@ -31,9 +34,8 @@
#include "Descriptor.h"
void
AmpStub::init (double _fs, bool adjust_downsampler)
AmpStub::init (bool adjust_downsampler)
{
fs = _fs;
dc_blocker.set_f (10. / fs);
/* going a bit lower than nominal with fc */
@@ -62,22 +64,20 @@ AmpStub::init (double _fs, bool adjust_downsampler)
s *= OVERSAMPLE;
for (int i = 0; i < up.n; ++i)
up.c[i] *= s;
normal = NOISE_FLOOR;
}
/* //////////////////////////////////////////////////////////////////////// */
void
AmpIII::init (double _fs)
AmpIII::init()
{
this->AmpStub::init (_fs, false);
this->AmpStub::init (false);
/* need to filter out dc before the power amp stage, which is running at
* the oversampled rate */
dc_blocker.set_f (10. / (fs * OVERSAMPLE));
DSP::RBJ::LoShelve (200 / (_fs), .2, -3, filter.a, filter.b);
DSP::RBJ::LoShelve (200 / fs, .2, -3, filter.a, filter.b);
}
template <sample_func_t F, int OVERSAMPLE>
@@ -86,10 +86,10 @@ AmpIII::one_cycle (int frames)
{
d_sample * s = ports[0];
d_sample gain = *ports[1];
d_sample temp = *ports[2] * tube.scale;
d_sample gain = getport(1);
d_sample temp = getport(2) * tube.scale;
drive = *ports[3] * .5;
drive = getport(3) * .5;
i_drive = 1 / (1 - drive);
d_sample * d = ports[4];
@@ -104,7 +104,7 @@ AmpIII::one_cycle (int frames)
/* recursive fade to prevent zipper noise from the 'gain' knob */
if (g == 0) g = current.g;
double one_over_n = 1. / frames;
double one_over_n = frames > 0 ? 1. / frames : 1;
double gf = pow (current.g / g, one_over_n);
for (int i = 0; i < frames; ++i)
@@ -123,14 +123,13 @@ AmpIII::one_cycle (int frames)
down.store (
power_transfer (
dc_blocker.process (
tube.transfer_clip (up.pad (o)))));
normal + tube.transfer_clip (up.pad (o)))));
F (d, i, a, adding_gain);
g *= gf;
}
normal = -normal;
current.g = g;
}
@@ -173,9 +172,9 @@ Descriptor<AmpIII>::setup()
Label = "AmpIII";
Properties = HARD_RT;
Name = "CAPS: AmpIII - Tube amp emulation";
Name = CAPS "AmpIII - Tube amp";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2002-5";
Copyright = "GPL, 2002-7";
/* fill port info and vtable */
autogen();
@@ -184,31 +183,31 @@ Descriptor<AmpIII>::setup()
/* //////////////////////////////////////////////////////////////////////// */
void
AmpIV::init (double _fs)
AmpIV::init()
{
this->AmpStub::init (_fs, false);
this->AmpStub::init (false);
/* need to filter out dc before the power amp stage, which is running at
* the oversampled rate */
dc_blocker.set_f (10. / (fs * OVERSAMPLE));
tone.init (_fs);
tone.init (fs);
}
template <sample_func_t F, int OVERSAMPLE>
void
AmpIV::one_cycle (int frames)
{
double one_over_n = 1. / frames;
double one_over_n = frames > 0 ? 1. / frames : 1;
d_sample * s = ports[0];
d_sample gain = *ports[1];
d_sample temp = *ports[2] * tube.scale;
d_sample gain = getport(1);
d_sample temp = getport(2) * tube.scale;
tone.start_cycle (ports + 3, one_over_n);
drive = *ports[7] * .5;
drive = getport(7) * .5;
i_drive = 1 / (1 - drive);
d_sample * d = ports[8];
@@ -241,14 +240,13 @@ AmpIV::one_cycle (int frames)
down.store (
power_transfer (
dc_blocker.process (
tube.transfer_clip (up.pad (o)))));
normal + tube.transfer_clip (up.pad (o)))));
F (d, i, a, adding_gain);
g *= gf;
}
normal = -normal;
current.g = g;
}
@@ -307,9 +305,9 @@ Descriptor<AmpIV>::setup()
Label = "AmpIV";
Properties = HARD_RT;
Name = "CAPS: AmpIV - Tube amp emulation + tone controls";
Name = CAPS "AmpIV - Tube amp + tone controls";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2002-5";
Copyright = "GPL, 2002-7";
/* fill port info and vtable */
autogen();
@@ -318,52 +316,50 @@ Descriptor<AmpIV>::setup()
/* //////////////////////////////////////////////////////////////////////// */
void
AmpV::init (double _fs)
AmpV::init()
{
this->AmpStub::init (_fs, false);
this->AmpStub::init (false);
/* need to filter out dc before the power amp stage, which is running at
* the oversampled rate */
dc_blocker.set_f (10. / (_fs * OVERSAMPLE));
dc_blocker.set_f (10. / (fs * OVERSAMPLE));
DSP::RBJ::LoShelve (210. / _fs, .2, -1, filter[0].a, filter[0].b);
DSP::RBJ::LoShelve (4200. / _fs, 1.2, +6, filter[1].a, filter[1].b);
DSP::RBJ::LoShelve (420. / _fs, .2, +2, filter[2].a, filter[2].b);
DSP::RBJ::LoShelve (210. / fs, .2, -1, filter[0].a, filter[0].b);
DSP::RBJ::LoShelve (4200. / fs, 1.2, +6, filter[1].a, filter[1].b);
DSP::RBJ::LoShelve (420. / fs, .2, +2, filter[2].a, filter[2].b);
/* power supply capacitor */
/* power supply cap */
for (int i = 0; i < 2; ++i)
DSP::RBJ::LP (10. / _fs, .3, power_cap[i].a, power_cap[i].b);
DSP::RBJ::LP (10. / fs, .3, power_cap[i].a, power_cap[i].b);
}
static int _turn = 0;
template <sample_func_t F, int OVERSAMPLE>
void
AmpV::one_cycle (int frames)
{
d_sample * s = ports[0];
d_sample gain = *ports[1];
d_sample gain = getport(1);
if (*ports[2] != cut)
{
cut = *ports[2];
cut = getport(2);
DSP::RBJ::LoShelve (210. / fs, .2, cut, filter[0].a, filter[0].b);
}
if (*ports[3] != tone)
{
tone = *ports[3];
tone = getport(3);
double f = tone * tone * 8400 + 420;
double q = tone * .4 + .2;
double db = tone * 2 + 2;
DSP::RBJ::LoShelve (f / fs, q, db, filter[2].a, filter[2].b);
}
drive = *ports[4] * .5;
drive = getport(4) * .5;
i_drive = 1 / (1 - drive);
#define MAX_WATTS port_info[5].range.UpperBound
d_sample sag = (MAX_WATTS - *ports[5]) / MAX_WATTS;
d_sample sag = (MAX_WATTS - getport(5)) / MAX_WATTS;
sag = .6 * sag * sag;
d_sample * d = ports[6];
@@ -373,13 +369,15 @@ AmpV::one_cycle (int frames)
double g = current.g;
current.g = max (gain < 1 ? gain : pow (20, gain - 1), .000001);
if (0 && (++_turn & 127) == 0)
#if 0
if (++_turn & 127) == 0)
fprintf (stderr, "supply = %.3f sag = %.3f\n", supply, sag);
#endif
if (g == 0) g = current.g;
/* recursive fade to prevent zipper noise from the 'gain' knob */
double one_over_n = 1. / frames;
double one_over_n = frames > 0 ? 1. / frames : 1;
double gf = pow (current.g / g, one_over_n);
for (int i = 0; i < frames; ++i)
@@ -409,7 +407,7 @@ AmpV::one_cycle (int frames)
down.store (
power_transfer (
dc_blocker.process (
tube.transfer_clip (
normal + tube.transfer_clip (
up.pad (o)))));
}
@@ -418,8 +416,8 @@ AmpV::one_cycle (int frames)
/* integrate for an approximation of cumulative output power */
supply += sag * fabs (a) + normal;
/* filter integrated power consumption */
for (int i = 0; i < 2; ++i)
supply = 0.9 * (power_cap[i].process (supply));
for (int j = 0; j < 2; ++j)
supply = 0.9 * (power_cap[j].process (supply));
g *= gf;
normal = -normal;
@@ -475,12 +473,165 @@ Descriptor<AmpV>::setup()
Label = "AmpV";
Properties = HARD_RT;
Name = "CAPS: AmpV - Refined tube amp emulation";
Name = CAPS "AmpV - Tube amp";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2002-5";
Copyright = "GPL, 2002-7";
/* fill port info and vtable */
autogen();
}
/* //////////////////////////////////////////////////////////////////////// */
void
AmpVTS::init()
{
this->AmpStub::init (false);
/* need to filter out dc before the power amp stage, which is running at
* the oversampled rate */
dc_blocker.set_f (10. / (fs * OVERSAMPLE));
/* power supply capacitance */
for (int i = 0; i < 2; ++i)
DSP::RBJ::LP (10. / fs, .3, power_cap[i].a, power_cap[i].b);
tonestack.init (fs);
}
template <sample_func_t F, int OVERSAMPLE>
void
AmpVTS::one_cycle (int frames)
{
d_sample * s = ports[0];
tonestack.start_cycle (ports + 1, 2);
d_sample gain = getport(2);
drive = getport(6) * .5;
i_drive = 1 / (1 - drive);
d_sample sag = 1 - max (0.0001, min (1, getport(7)));
sag = .6 * sag * sag; /* map to log space makes slider better */
d_sample * d = ports[8];
*ports[9] = OVERSAMPLE;
double g = current.g;
if (gain < 1)
current.g = max (gain, .001);
else
{
gain -= 1;
gain *= gain;
current.g = pow (10, gain);
}
/* recursive fade to prevent zipper noise from the 'gain' knob */
double one_over_n = frames > 0 ? 1. / frames : 1;
double gf = pow (current.g / g, one_over_n);
for (int i = 0; i < frames; ++i)
{
register double a = s[i];
register double v = 3 - supply;
v = v * v * .06 + .46;
a = tube.transfer (a);
a = tonestack.process (a + normal);
a = g * (a + supply * .001);
a = v * tube.transfer_clip (up.upsample (a));
a = power_transfer (dc_blocker.process (a));
a = down.process (a);
{
for (int o = 1; o < OVERSAMPLE; ++o)
down.store (
power_transfer (
dc_blocker.process (
normal + tube.transfer_clip (
up.pad (o)))));
}
F (d, i, a, adding_gain);
/* integrate for an approximation of cumulative output power */
supply += sag * fabs (a) + normal;
/* filter integrated power consumption */
for (int j = 0; j < 2; ++j)
supply = 0.9 * (power_cap[j].process (supply + normal));
g *= gf;
normal = -normal;
}
current.g = g;
}
/* //////////////////////////////////////////////////////////////////////// */
PortInfo
AmpVTS::port_info [] =
{
{
"in",
INPUT | AUDIO,
{BOUNDED, -1, 1}
}, {
"model",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0 | INTEGER, 0, 5} /* no way to set dyn at compile t */
}, {
"gain",
INPUT | CONTROL,
{BOUNDED | DEFAULT_HIGH, 0, 3}
}, {
"bass",
INPUT | CONTROL,
{BOUNDED | DEFAULT_MID, 0, 1}
}, {
"mid",
INPUT | CONTROL,
{BOUNDED | DEFAULT_1, 0, 1}
}, {
"treble",
INPUT | CONTROL,
{BOUNDED | DEFAULT_HIGH, 0, 1}
}, {
"drive",
INPUT | CONTROL,
{BOUNDED | DEFAULT_LOW, 0.0001, 1}
}, {
"watts",
INPUT | CONTROL,
{BOUNDED | DEFAULT_HIGH, 0.0001, 1}
}, {
"out",
OUTPUT | AUDIO,
{0}
}, {
"latency",
OUTPUT | CONTROL,
{0}
}
};
template <> void
Descriptor<AmpVTS>::setup()
{
UniqueID = 2592;
Label = "AmpVTS";
Properties = HARD_RT;
Name = CAPS "AmpVTS - Tube amp + Tone stack";
Maker = "David Yeh <dtyeh@ccrma.stanford.edu> & Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2002-7";
/* fill port info and vtable */
autogen();
}

96
plugins/ladspa_effect/caps/Amp.h
View File

@@ -1,11 +1,11 @@
/*
Amp.h
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
oversampled tube amplifier emulation.
Oversampled tube amplifier emulation.
*/
/*
@@ -41,14 +41,12 @@
#include "dsp/RBJ.h"
#include "dsp/Eq.h"
#include "dsp/ToneStack.h"
class AmpStub
: public Plugin
{
public:
double fs;
/* oscillating NOISE_FLOOR, added to prevent denormals in signal */
d_sample normal;
DSP::TwelveAX7_3 tube;
d_sample drive, i_drive;
@@ -78,7 +76,7 @@ class AmpStub
down (FIR_SIZE, up.c)
{ }
void init (double _fs, bool adjust_downsampler = false);
void init (bool adjust_downsampler = false);
inline d_sample power_transfer (d_sample a)
{
@@ -93,18 +91,16 @@ class PreampIII
{
public:
template <sample_func_t F, int OVERSAMPLE>
void one_cycle (int frames);
void one_cycle (int frames);
DSP::BiQuad filter;
public:
static PortInfo port_info[];
d_sample * ports [5];
d_sample adding_gain;
void init (double _fs);
void init();
void activate()
{
current.g = 1;
@@ -133,18 +129,16 @@ class AmpIII
{
public:
template <sample_func_t F, int OVERSAMPLE>
void one_cycle (int frames);
void one_cycle (int frames);
DSP::BiQuad filter;
public:
static PortInfo port_info[];
d_sample * ports [6];
d_sample adding_gain;
void init (double _fs);
void init();
void activate()
{
current.g = 1;
@@ -176,9 +170,8 @@ class ToneControls
{
public:
d_sample eq_gain[4];
DSP::Eq<4,4> eq;
DSP::Eq<4> eq;
static PreampBand bands[4];
d_sample normal;
public:
void init (double _fs);
@@ -220,16 +213,14 @@ class PreampIV
ToneControls tone;
template <sample_func_t F, int OVERSAMPLE>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info[];
d_sample * ports [9];
d_sample adding_gain;
void init (double _fs);
void init();
void activate();
void run (int n)
@@ -252,16 +243,14 @@ class AmpIV
ToneControls tone;
template <sample_func_t F, int OVERSAMPLE>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info[];
d_sample * ports [10];
d_sample adding_gain;
void init (double _fs);
void init();
void activate()
{
current.g = 1;
@@ -291,7 +280,7 @@ class AmpV
{
public:
template <sample_func_t F, int OVERSAMPLE>
void one_cycle (int frames);
void one_cycle (int frames);
DSP::BiQuad filter[3];
@@ -303,12 +292,10 @@ class AmpV
public:
static PortInfo port_info[];
d_sample * ports [7];
d_sample adding_gain;
void init (double _fs);
void init();
void activate()
{
current.g = 1;
@@ -336,4 +323,53 @@ class AmpV
}
};
/* /////////////////////////////////////////////////////////////////////// */
class AmpVTS
: public AmpStub
{
public:
DSP::ToneStack tonestack;
template <sample_func_t F, int OVERSAMPLE>
void one_cycle (int frames);
d_sample cut, tone;
/* supply voltage sag */
d_sample supply;
DSP::BiQuad power_cap[2];
public:
static PortInfo port_info[];
d_sample adding_gain;
void init();
void activate()
{
current.g = 1;
for (int i = 0; i < 2; ++i)
power_cap[i].reset();
up.reset();
down.reset();
dc_blocker.reset();
cut = 2;
supply = 0.;
}
void run (int n)
{
one_cycle<store_func, OVERSAMPLE> (n);
}
void run_adding (int n)
{
one_cycle<adding_func, OVERSAMPLE> (n);
}
};
#endif /* _AMP_H_ */

37
plugins/ladspa_effect/caps/Cabinet.cc
View File

@@ -1,7 +1,7 @@
/*
Cabinet.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -73,11 +73,10 @@ CabinetI::models [] =
/* //////////////////////////////////////////////////////////////////////// */
void
CabinetI::init (double fs)
CabinetI::init()
{
h = 0;
model = 0;
normal = NOISE_FLOOR;
}
void
@@ -92,7 +91,7 @@ CabinetI::switch_model (int m)
a = models[m].a;
b = models[m].b;
gain = models[m].gain * DSP::db2lin (*ports[2]);
gain = models[m].gain * DSP::db2lin (getport(2));
memset (x, 0, sizeof (x));
memset (y, 0, sizeof (y));
@@ -101,7 +100,7 @@ CabinetI::switch_model (int m)
void
CabinetI::activate()
{
switch_model ((int) *ports[1]);
switch_model ((int) getport(1));
}
template <sample_func_t F>
@@ -110,10 +109,10 @@ CabinetI::one_cycle (int frames)
{
d_sample * s = ports[0];
int m = (int) *ports[1];
int m = (int) getport (1);
if (m != model) switch_model (m);
d_sample g = models[model].gain * DSP::db2lin (*ports[2]);
d_sample g = models[model].gain * DSP::db2lin (getport(2));
double gf = pow (g / gain, 1 / (double) frames);
d_sample * d = ports[3];
@@ -140,8 +139,6 @@ CabinetI::one_cycle (int frames)
F (d, i, gain * out, adding_gain);
gain *= gf;
}
normal = -normal;
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -177,9 +174,9 @@ Descriptor<CabinetI>::setup()
Label = "CabinetI";
Properties = HARD_RT;
Name = "CAPS: CabinetI - Loudspeaker cabinet emulation";
Name = CAPS "CabinetI - Loudspeaker cabinet emulation";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2002-5";
Copyright = "GPL, 2002-7";
/* fill port info and vtable */
autogen();
@@ -190,7 +187,7 @@ Descriptor<CabinetI>::setup()
#include "Cabinet-Models32.h"
void
CabinetII::init (double fs)
CabinetII::init()
{
if (fs < 46000)
models = models44100;
@@ -203,7 +200,6 @@ CabinetII::init (double fs)
h = 0;
model = 0;
normal = NOISE_FLOOR;
}
void
@@ -218,7 +214,7 @@ CabinetII::switch_model (int m)
a = models[m].a;
b = models[m].b;
gain = models[m].gain * DSP::db2lin (*ports[2]);
gain = models[m].gain * DSP::db2lin (getport(2));
memset (x, 0, sizeof (x));
memset (y, 0, sizeof (y));
@@ -227,8 +223,7 @@ CabinetII::switch_model (int m)
void
CabinetII::activate()
{
switch_model ((int) *ports[1]);
gain = models[model].gain * DSP::db2lin (*ports[2]);
switch_model ((int) getport(1));
}
template <sample_func_t F>
@@ -237,10 +232,10 @@ CabinetII::one_cycle (int frames)
{
d_sample * s = ports[0];
int m = (int) *ports[1];
int m = (int) getport (1);
if (m != model) switch_model (m);
d_sample g = models[model].gain * DSP::db2lin (*ports[2]);
d_sample g = models[model].gain * DSP::db2lin (getport(2));
double gf = pow (g / gain, 1 / (double) frames);
d_sample * d = ports[3];
@@ -267,8 +262,6 @@ CabinetII::one_cycle (int frames)
F (d, i, gain * out, adding_gain);
gain *= gf;
}
normal = -normal;
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -304,9 +297,9 @@ Descriptor<CabinetII>::setup()
Label = "CabinetII";
Properties = HARD_RT;
Name = "CAPS: CabinetII - Refined loudspeaker cabinet emulation";
Name = CAPS "CabinetII - Refined loudspeaker cabinet emulation";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2002-5";
Copyright = "GPL, 2002-7";
/* fill port info and vtable */
autogen();

19
plugins/ladspa_effect/caps/Cabinet.h
View File

@@ -54,6 +54,7 @@ typedef struct {
} Model32;
class CabinetI
: public Plugin
{
public:
d_sample gain;
@@ -66,18 +67,13 @@ class CabinetI
cabinet_float * a, * b;
cabinet_float x[16], y[16];
d_sample normal;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [4];
d_sample adding_gain;
void init (double _fs);
void init();
void activate();
@@ -96,6 +92,7 @@ class CabinetI
* 44.1 / 48 / 88.2 / 96 kHz sample rates */
class CabinetII
: public Plugin
{
public:
d_sample gain;
@@ -113,19 +110,15 @@ class CabinetII
cabinet_float * a, * b;
cabinet_float x[32], y[32];
d_sample normal;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [4];
d_sample adding_gain;
void init (double _fs);
void init();
void activate();
void run (int n)

76
plugins/ladspa_effect/caps/Chorus.cc
View File

@@ -1,7 +1,7 @@
/*
Chorus.cc
Copyright 2004, 2005 Tim Goetze <tim@quitte.de>
Copyright 2004-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -40,27 +40,24 @@ ChorusI::one_cycle (int frames)
double ms = .001 * fs;
double t = time;
time = *ports[1] * ms;
time = getport(1) * ms;
double dt = (time - t) * one_over_n;
double w = width;
width = *ports[2] * ms;
width = getport(2) * ms;
/* clamp, or we need future samples from the delay line */
if (width >= t - 3) width = t - 3;
double dw = (width - w) * one_over_n;
if (rate != *ports[3])
lfo.set_f (max (rate = *ports[3], .000001), fs, lfo.get_phase());
lfo.set_f (max (rate = getport(3), .000001), fs, lfo.get_phase());
double blend = *ports[4];
double ff = *ports[5];
double fb = *ports[6];
double blend = getport(4);
double ff = getport(5);
double fb = getport(6);
d_sample * d = ports[7];
/* flip 'renormal' addition constant */
normal = -normal;
DSP::FPTruncateMode truncate;
for (int i = 0; i < frames; ++i)
@@ -142,9 +139,9 @@ Descriptor<ChorusI>::setup()
Label = "ChorusI";
Properties = HARD_RT;
Name = "CAPS: ChorusI - Mono chorus/flanger";
Name = CAPS "ChorusI - Mono chorus/flanger";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
@@ -162,34 +159,31 @@ StereoChorusI::one_cycle (int frames)
double ms = .001 * fs;
double t = time;
time = *ports[1] * ms;
time = getport(1) * ms;
double dt = (time - t) * one_over_n;
double w = width;
width = *ports[2] * ms;
width = getport(2) * ms;
/* clamp, or we need future samples from the delay line */
if (width >= t - 1) width = t - 1;
double dw = (width - w) * one_over_n;
if (rate != *ports[3] && phase != *ports[4])
{
rate = *ports[3];
phase = *ports[4];
rate = getport(3);
phase = getport(4);
double phi = left.lfo.get_phase();
left.lfo.set_f (max (rate, .000001), fs, phi);
right.lfo.set_f (max (rate, .000001), fs, phi + phase * M_PI);
}
double blend = *ports[5];
double ff = *ports[6];
double fb = *ports[7];
double blend = getport(5);
double ff = getport(6);
double fb = getport(7);
d_sample * dl = ports[8];
d_sample * dr = ports[9];
/* flip 'renormal' addition constant */
normal = -normal;
/* to go sure (on i386) that the fistp instruction does the right thing
* when looking up fractional sample indices */
DSP::FPTruncateMode truncate;
@@ -273,9 +267,9 @@ Descriptor<StereoChorusI>::setup()
Label = "StereoChorusI";
Properties = HARD_RT;
Name = "CAPS: StereoChorusI - Stereo chorus/flanger";
Name = CAPS "StereoChorusI - Stereo chorus/flanger";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
@@ -293,11 +287,11 @@ ChorusII::one_cycle (int frames)
double ms = .001 * fs;
double t = time;
time = *ports[1] * ms;
time = getport(1) * ms;
double dt = (time - t) * one_over_n;
double w = width;
width = *ports[2] * ms;
width = getport(2) * ms;
/* clamp, or we need future samples from the delay line */
if (width >= t - 3) width = t - 3;
double dw = (width - w) * one_over_n;
@@ -305,15 +299,12 @@ ChorusII::one_cycle (int frames)
if (rate != *ports[3])
set_rate (*ports[3]);
double blend = *ports[4];
double ff = *ports[5];
double fb = *ports[6];
double blend = getport(4);
double ff = getport(5);
double fb = getport(6);
d_sample * d = ports[7];
/* flip 'renormal' addition constant */
normal = -normal;
DSP::FPTruncateMode truncate;
for (int i = 0; i < frames; ++i)
@@ -384,9 +375,9 @@ Descriptor<ChorusII>::setup()
Label = "ChorusII";
Properties = HARD_RT;
Name = "CAPS: ChorusII - Mono chorus/flanger modulated by a fractal";
Name = CAPS "ChorusII - Mono chorus/flanger modulated by a fractal";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
@@ -404,27 +395,24 @@ StereoChorusII::one_cycle (int frames)
double ms = .001 * fs;
double t = time;
time = *ports[1] * ms;
time = getport(1) * ms;
double dt = (time - t) * one_over_n;
double w = width;
width = *ports[2] * ms;
width = getport(2) * ms;
/* clamp, or we need future samples from the delay line */
if (width >= t - 1) width = t - 1;
double dw = (width - w) * one_over_n;
set_rate (*ports[3]);
double blend = *ports[4];
double ff = *ports[5];
double fb = *ports[6];
double blend = getport(4);
double ff = getport(5);
double fb = getport(6);
d_sample * dl = ports[7];
d_sample * dr = ports[8];
/* flip 'renormal' addition constant */
normal = -normal;
/* to go sure (on i386) that the fistp instruction does the right thing
* when looking up fractional sample indices */
DSP::FPTruncateMode truncate;
@@ -507,9 +495,9 @@ Descriptor<StereoChorusII>::setup()
Label = "StereoChorusII";
Properties = HARD_RT;
Name = "CAPS: StereoChorusII - Stereo chorus/flanger modulated by a fractal";
Name = CAPS "StereoChorusII - Stereo chorus/flanger modulated by a fractal";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

38
plugins/ladspa_effect/caps/Chorus.h
View File

@@ -38,11 +38,10 @@
#include "dsp/RBJ.h"
class ChorusStub
: public Plugin
{
public:
double fs;
d_sample time, width, rate;
d_sample normal; /* denormal protection */
};
class ChorusI
@@ -54,20 +53,15 @@ class ChorusI
DSP::DelayTapA tap;
template <sample_func_t>
void one_cycle (int frames);
void one_cycle (int frames);
public:
d_sample * ports [8];
static PortInfo port_info [];
d_sample adding_gain;
void init (double _fs)
void init()
{
fs = _fs;
rate = .15;
delay.init ((int) (.040 * fs));
normal = NOISE_FLOOR;
}
void activate()
@@ -98,7 +92,6 @@ class StereoChorusI
: public ChorusStub
{
public:
double fs;
d_sample rate;
d_sample phase;
@@ -110,22 +103,17 @@ class StereoChorusI
} left, right;
template <sample_func_t>
void one_cycle (int frames);
void one_cycle (int frames);
public:
d_sample * ports [10];
static PortInfo port_info [];
d_sample adding_gain;
void init (double _fs)
void init()
{
fs = _fs;
rate = .15;
phase = .5; /* pi */
delay.init ((int) (.040 * fs));
normal = NOISE_FLOOR;
}
void activate()
@@ -200,7 +188,7 @@ class ChorusII
DSP::Delay delay;
template <sample_func_t>
void one_cycle (int frames);
void one_cycle (int frames);
void set_rate (d_sample r)
{
@@ -213,15 +201,10 @@ class ChorusII
}
public:
d_sample * ports [8];
static PortInfo port_info [];
d_sample adding_gain;
void init (double _fs)
void init()
{
fs = _fs;
normal = NOISE_FLOOR;
delay.init ((int) (.040 * fs));
for (int i = 0; i < Taps; ++i)
taps[i].init (fs);
@@ -254,7 +237,6 @@ class StereoChorusII
: public ChorusStub
{
public:
double fs;
d_sample rate;
d_sample phase;
@@ -279,18 +261,14 @@ class StereoChorusII
}
public:
d_sample * ports [10];
static PortInfo port_info [];
d_sample adding_gain;
void init (double _fs)
void init()
{
fs = _fs;
phase = .5; /* pi */
delay.init ((int) (.040 * fs));
normal = NOISE_FLOOR;
left.fractal.init (.001, frandom());
right.fractal.init (.001, frandom());

38
plugins/ladspa_effect/caps/Click.cc
View File

@@ -1,7 +1,7 @@
/*
Click.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -27,28 +27,26 @@
#include "basics.h"
#include "click.h"
#include "money.h"
#include "waves/click.h"
#include "waves/money.h"
#include "Click.h"
#include "Descriptor.h"
void
ClickStub::init (double _fs, float * _wave, int _N)
ClickStub::init (float * _wave, int _N)
{
fs = _fs;
wave = _wave;
N = _N;
bpm = -1;
normal = NOISE_FLOOR;
}
template <sample_func_t F>
void
ClickStub::one_cycle (int frames)
{
bpm = *ports[0];
d_sample gain = *ports[1] * *ports[1];
bpm = getport(0);
d_sample gain = getport(1) * *ports[1];
lp.set (1 - *ports[2]);
d_sample * d = ports[3];
@@ -120,9 +118,9 @@ ClickStub::port_info [] =
#define LENGTH(W) ((int) (sizeof (W) / sizeof (float)))
void
Click::init (double fs)
Click::init()
{
this->ClickStub::init (fs, click, LENGTH (click));
this->ClickStub::init (click, LENGTH (click));
}
template <> void
@@ -132,9 +130,9 @@ Descriptor<Click>::setup()
Label = "Click";
Properties = HARD_RT;
Name = "CAPS: Click - Metronome";
Name = CAPS "Click - Metronome";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
@@ -165,9 +163,9 @@ CEO::port_info [] =
};
void
CEO::init (double fs)
CEO::init()
{
this->ClickStub::init (fs, money, LENGTH (money));
this->ClickStub::init (money, LENGTH (money));
}
template <> void
@@ -177,9 +175,9 @@ Descriptor<CEO>::setup()
Label = "CEO";
Properties = HARD_RT;
Name = "CAPS: CEO - Chief Executive Oscillator";
Name = CAPS "CEO - Chief Executive Oscillator";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
@@ -212,9 +210,9 @@ Dirac::port_info [] =
};
void
Dirac::init (double fs)
Dirac::init()
{
this->ClickStub::init (fs, dirac, LENGTH (dirac));
this->ClickStub::init (dirac, LENGTH (dirac));
}
template <> void
@@ -224,9 +222,9 @@ Descriptor<Dirac>::setup()
Label = "Dirac";
Properties = HARD_RT;
Name = "CAPS: Dirac - One-sample impulse generator";
Name = CAPS "Dirac - One-sample impulse generator";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

18
plugins/ladspa_effect/caps/Click.h
View File

@@ -32,10 +32,9 @@
#include "dsp/util.h"
class ClickStub
: public Plugin
{
public:
double fs;
d_sample bpm;
float * wave;
@@ -46,18 +45,13 @@ class ClickStub
int period; /* frames remaining in period */
int played; /* frames played from sample */
d_sample normal;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [4];
d_sample adding_gain;
void init (double _fs, float * _wave, int _N);
void init (float * _wave, int _N);
void activate()
{
@@ -80,14 +74,14 @@ class Click
: public ClickStub
{
public:
void init (double fs);
void init();
};
class CEO
: public ClickStub
{
public:
void init (double fs);
void init();
static PortInfo port_info [];
};
@@ -96,7 +90,7 @@ class Dirac
: public ClickStub
{
public:
void init (double fs);
void init();
static PortInfo port_info [];
};

14
plugins/ladspa_effect/caps/Clip.cc
View File

@@ -1,7 +1,7 @@
/*
Clip.cc
Copyright 2003-5 Tim Goetze <tim@quitte.de>
Copyright 2003-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -31,9 +31,8 @@
#include "Descriptor.h"
void
Clip::init (double _fs)
Clip::init()
{
fs = _fs;
gain = 1;
threshold[0] = -.9;
@@ -79,12 +78,13 @@ Clip::one_cycle (int frames)
{
d_sample * s = ports[0];
double g = getport (1);
double gf;
if (*ports[1] == gain_db)
if (g == gain_db)
gf = 1;
else
{
gain_db = *ports[1];
gain_db = g;
d_sample g = DSP::db2lin (gain_db);
gf = pow (g / gain, 1 / (double) frames);
}
@@ -138,9 +138,9 @@ Descriptor<Clip>::setup()
Label = "Clip";
Properties = HARD_RT;
Name = "CAPS: Clip - Hard clipper, 8x oversampled";
Name = CAPS "Clip - Hard clipper, 8x oversampled";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2003-5";
Copyright = "GPL, 2003-7";
/* fill port info and vtable */
autogen();

9
plugins/ladspa_effect/caps/Clip.h
View File

@@ -34,9 +34,9 @@
#include "dsp/windows.h"
class Clip
: public Plugin
{
public:
double fs;
d_sample gain, gain_db;
d_sample threshold[2];
@@ -51,22 +51,19 @@ class Clip
DSP::FIR down;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
inline d_sample clip (d_sample x);
public:
static PortInfo port_info[];
d_sample * ports [4];
d_sample adding_gain;
Clip()
: up (FIR_SIZE, OVERSAMPLE),
down (FIR_SIZE)
{ }
void init (double fs);
void init();
void activate()
{

24
plugins/ladspa_effect/caps/Compress.cc
View File

@@ -1,7 +1,7 @@
/*
Compress.cc
Copyright 2004-5 Tim Goetze <tim@quitte.de>
Copyright 2004-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -31,20 +31,14 @@
#include "Compress.h"
#include "Descriptor.h"
void
Compress::init (double _fs)
{
fs = _fs;
}
template <sample_func_t F>
void
Compress::one_cycle (int frames)
{
d_sample * s = ports[0];
d_sample range = DSP::db2lin (*ports[1]);
d_sample ratio = (*ports[2] - 1) / *ports[2];
d_sample range = DSP::db2lin (getport(1));
d_sample ratio = (*ports[2] - 1) / getport(2);
/* sc1 has lookup tables here, and they're only 40 % used (400 ms/1 s).
* thus, sc1's attack/release controls are a bit coarse due to truncation
@@ -57,11 +51,11 @@ Compress::one_cycle (int frames)
*
* TODO: check whether these parameters work like they should, try pow()
*/
double ga = exp (-1 / (fs * *ports[3]));
double gr = exp (-1 / (fs * *ports[4]));
double ga = exp (-1 / (fs * getport(3)));
double gr = exp (-1 / (fs * getport(4)));
d_sample threshold = *ports[5];
d_sample knee = *ports[6];
d_sample threshold = getport(5);
d_sample knee = getport(6);
d_sample * d = ports[7];
@@ -149,9 +143,9 @@ Descriptor<Compress>::setup()
Label = "Compress";
Properties = HARD_RT;
Name = "CAPS: Compress - Mono compressor";
Name = CAPS "Compress - Mono compressor";
Maker = "Tim Goetze <tim@quitte.de>, Steve Harris <steve@plugin.org.uk>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

9
plugins/ladspa_effect/caps/Compress.h
View File

@@ -32,6 +32,7 @@
#include "dsp/util.h"
class Compress
: public Plugin
{
public:
double fs;
@@ -43,16 +44,12 @@ class Compress
int count;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [8];
d_sample adding_gain;
void init (double _fs);
void init() {}
void activate()
{
rms.reset();

125
plugins/ladspa_effect/caps/Descriptor.h
View File

@@ -1,24 +1,18 @@
/*
Descriptor.h
Copyright 2004-5 Tim Goetze <tim@quitte.de>
Copyright 2004-6 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
creating a LADSPA_Descriptor for a caps plugin via a C++ template,
Creating a LADSPA_Descriptor for a CAPS plugin via a C++ template,
saves us a virtual function call compared to the usual method used
for C++ plugins in a C context.
Descriptor<P> expects P to declare some common methods, like init(),
activate() etc, plus a static port_info[] and LADSPA_Data * ports[]
and of course 'adding_gain'.
maintaining both port_info[] and ports[] is a bit of a bitch, but,
hey, "you only do it once (tm)" .. and then you do it over and over
again. particularly bothersome is also the necessary unrolling of our
PortInfo array to fit into LADSPA_Descriptor's inconsequential way of
port data structuring, which results in quite a bit of memory holding
duplicated data. oh well.
and adding_gain.
*/
/*
This program is free software; you can redistribute it and/or
@@ -40,6 +34,13 @@
#ifndef _DESCRIPTOR_H_
#define _DESCRIPTOR_H_
#ifdef __SSE__
#include <xmmintrin.h>
#endif
#ifdef __SSE3__
#include <pmmintrin.h>
#endif
/* common stub for Descriptor makes it possible to delete() without special-
* casing for every plugin class.
*/
@@ -47,6 +48,8 @@ class DescriptorStub
: public LADSPA_Descriptor
{
public:
static int thishostsucks;
DescriptorStub()
{
PortCount = 0;
@@ -62,23 +65,34 @@ class DescriptorStub
}
}
};
inline void
processor_specific_denormal_measures()
{
#ifdef __SSE3__
/* this one works reliably on a 6600 Core2 */
_MM_SET_DENORMALS_ZERO_MODE (_MM_DENORMALS_ZERO_ON);
#endif
#ifdef __SSE__
/* this one doesn't ... */
_MM_SET_FLUSH_ZERO_MODE (_MM_FLUSH_ZERO_ON);
#endif
}
template <class T>
class Descriptor
: public DescriptorStub
{
public:
/* tom szilyagi reports that hosts exist which call activate() before
* connect_port(). since caps' plugins expect ports to be valid we
/* Tom Szilyagi reports that hosts exist which call activate() before
* connect_port(). Since CAPS' plugins expect ports to be valid we
* need a safeguard: at instantiation, each port is connected to the
* lower bound. When (If?) LADSPA default values are ever fixed, connecting
* to the default will be preferred. */
* lower bound. */
LADSPA_PortRangeHint * ranges;
public:
Descriptor()
{ setup(); }
Descriptor() { setup(); }
void setup();
void autogen()
@@ -117,12 +131,20 @@ class Descriptor
const struct _LADSPA_Descriptor * d, ulong fs)
{
T * plugin = new T();
/* see comment above at 'ranges' member */
for (int i = 0; i < (int) d->PortCount; ++i)
plugin->ports[i] = &((Descriptor *) d)->ranges[i].LowerBound;
int n = (int) d->PortCount;
plugin->init (fs);
LADSPA_PortRangeHint * ranges = ((Descriptor *) d)->ranges;
plugin->ranges = ranges;
plugin->ports = new d_sample * [n];
/* connect to lower bound as a safety measure */
for (int i = 0; i < n; ++i)
plugin->ports[i] = &(ranges[i].LowerBound);
plugin->fs = fs;
plugin->normal = NOISE_FLOOR;
plugin->init();
return plugin;
}
@@ -134,29 +156,72 @@ class Descriptor
static void _activate (LADSPA_Handle h)
{
((T *) h)->activate();
T * plugin = (T *) h;
plugin->first_run = 1;
/* since none of the plugins do any RT-critical work in
* activate(), it's safe to defer the actual call to the
* plugin's activate() method for the first run() after
* the host called in here.
*
* It's simplest way to prevent a parameter smoothing sweep
* in the first audio block after activation.
plugin->activate();
*/
}
static void _run (LADSPA_Handle h, ulong n)
{
/* cannot call template here (g++ 2.95), sigh. */
((T *) h)->run (n);
T * plugin = (T *) h;
/* We don't reset the processor flags later, it's true. */
processor_specific_denormal_measures();
/* If this is the first audio block after activation,
* initialize the plugin from the current set of parameters. */
if (plugin->first_run)
{
plugin->activate();
plugin->first_run = 0;
}
plugin->run (n);
plugin->normal = -plugin->normal;
}
static void _run_adding (LADSPA_Handle h, ulong n)
{
/* cannot call a template here (g++ 2.95), sigh. */
((T *) h)->run_adding (n);
T * plugin = (T *) h;
/* We don't reset the processor flags later, it's true. */
processor_specific_denormal_measures();
/* If this is the first audio block after activation,
* initialize the plugin from the current set of parameters. */
if (plugin->first_run)
{
plugin->activate();
plugin->first_run = 0;
}
plugin->run_adding (n);
plugin->normal = -plugin->normal;
}
static void _set_run_adding_gain (LADSPA_Handle h, LADSPA_Data g)
{
((T *) h)->adding_gain = g;
T * plugin = (T *) h;
plugin->adding_gain = g;
}
static void _cleanup (LADSPA_Handle h)
{
delete (T *) h;
T * plugin = (T *) h;
delete [] plugin->ports;
delete plugin;
}
};

219
plugins/ladspa_effect/caps/Eq.cc
View File

@@ -1,7 +1,7 @@
/*
Eq.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -31,14 +31,8 @@
#include "Eq.h"
#include "Descriptor.h"
void
Eq::init (double _fs)
{
fs = _fs;
eq.init (fs, 1.2);
normal = NOISE_FLOOR;
}
/* slight adjustments to gain to keep response optimally flat at
* 0 dB gain in all bands */
inline static double
adjust_gain (int i, double g)
{
@@ -53,13 +47,22 @@ adjust_gain (int i, double g)
return g * adjust[i];
}
#define Q 1.2
void
Eq::init()
{
eq.init (fs, Q);
}
void
Eq::activate()
{
for (int i = 0; i < 10; ++i)
{
gain[i] = *ports [1 + i];
gain[i] = getport (1 + i);
eq.gain[i] = adjust_gain (i, DSP::db2lin (gain[i]));
eq.gf[i] = 1;
}
}
@@ -71,29 +74,34 @@ Eq::one_cycle (int frames)
/* evaluate band gain changes and compute recursion factor to prevent
* zipper noise */
double one_over_n = 1. / frames;
double one_over_n = frames > 0 ? 1. / frames : 1;
for (int i = 0; i < 10; ++i)
{
if (*ports [1 + i] == gain[i])
d_sample g = getport (1 + i);
if (g == gain[i])
{
/* no gain factoring */
eq.gf[i] = 1;
continue;
}
gain[i] = g;
gain[i] = *ports [1 + i];
double want = adjust_gain (i, DSP::db2lin (gain[i]));
double want = adjust_gain (i, DSP::db2lin (g));
eq.gf[i] = pow (want / eq.gain[i], one_over_n);
}
d_sample * d = ports[11];
for (int i = 0; i < frames; ++i)
F (d, i, eq.process (s[i] + normal), adding_gain);
{
d_sample x = s[i];
x = eq.process (x);
F (d, i, x, adding_gain);
}
normal = -normal;
eq.normal = -normal;
eq.flush_0();
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -108,43 +116,43 @@ Eq::port_info [] =
}, {
"31 Hz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 30}
{BOUNDED | DEFAULT_LOW, -48, 24}
}, {
"63 Hz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 30}
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"125 Hz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 30}
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"250 Hz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 30}
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"500 Hz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 30}
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"1 kHz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 30}
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"2 kHz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 30}
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"4 kHz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 30}
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"8 kHz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 30}
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"16 kHz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 30}
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"out",
OUTPUT | AUDIO,
@@ -159,11 +167,164 @@ Descriptor<Eq>::setup()
Label = "Eq";
Properties = HARD_RT;
Name = "CAPS: Eq - 10-band 'analogue' equalizer";
Name = CAPS "Eq - 10-band equalizer";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
}
/* //////////////////////////////////////////////////////////////////////// */
void
Eq2x2::init()
{
for (int c = 0; c < 2; ++c)
eq[c].init (fs, Q);
}
void
Eq2x2::activate()
{
/* Fetch current parameter settings so we won't sweep band gains in the
* first block to process.
*/
for (int i = 0; i < 10; ++i)
{
gain[i] = getport (2 + i);
double a = adjust_gain (i, DSP::db2lin (gain[i]));
for (int c = 0; c < 2; ++c)
eq[c].gf[i] = 1,
eq[c].gain[i] = a;
}
}
template <sample_func_t F>
void
Eq2x2::one_cycle (int frames)
{
/* evaluate band gain changes and compute recursion factor to prevent
* zipper noise */
double one_over_n = frames > 0 ? 1. / frames : 1;
for (int i = 0; i < 10; ++i)
{
double a;
if (*ports [2 + i] == gain[i])
/* still same value, no gain fade */
a = 1;
else
{
gain[i] = getport (2 + i);
/* prepare factor for logarithmic gain fade */
a = adjust_gain (i, DSP::db2lin (gain[i]));
a = pow (a / eq[0].gain[i], one_over_n);
}
for (int c = 0; c < 2; ++c)
eq[c].gf[i] = a;
}
for (int c = 0; c < 2; ++c)
{
d_sample
* s = ports[c],
* d = ports[12 + c];
for (int i = 0; i < frames; ++i)
{
d_sample x = s[i];
x = eq[c].process (x);
F (d, i, x, adding_gain);
}
}
/* flip 'renormal' values */
for (int c = 0; c < 2; ++c)
{
eq[c].normal = normal;
eq[c].flush_0();
}
}
PortInfo
Eq2x2::port_info [] =
{
{
"in:l",
INPUT | AUDIO,
{0, -1, 1}
}, {
"in:r",
INPUT | AUDIO,
{0, -1, 1}
}, {
"31 Hz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"63 Hz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"125 Hz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"250 Hz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"500 Hz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"1 kHz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"2 kHz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"4 kHz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"8 kHz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"16 kHz",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0, -48, 24}
}, {
"out:l",
OUTPUT | AUDIO,
{0}
}, {
"out:r",
OUTPUT | AUDIO,
{0}
}
};
template <> void
Descriptor<Eq2x2>::setup()
{
UniqueID = 2594;
Label = "Eq2x2";
Properties = HARD_RT;
Name = CAPS "Eq2x2 - stereo 10-band equalizer";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
}

42
plugins/ladspa_effect/caps/Eq.h
View File

@@ -28,29 +28,55 @@
#ifndef _EQ_H_
#define _EQ_H_
#include "dsp/Eq.h"
#include "dsp/util.h"
#include "dsp/Eq.h"
#include "dsp/BiQuad.h"
#include "dsp/RBJ.h"
class Eq
: public Plugin
{
public:
double fs;
d_sample gain[10];
d_sample normal;
DSP::Eq<10> eq;
DSP::Eq<10,12> eq;
int block;
enum { BlockSize = 64 };
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [12];
d_sample adding_gain;
void init();
void activate();
void init (double _fs);
void run (int n)
{
one_cycle<store_func> (n);
}
void run_adding (int n)
{
one_cycle<adding_func> (n);
}
};
class Eq2x2
: public Plugin
{
public:
d_sample gain[10];
DSP::Eq<10> eq[2];
template <sample_func_t F>
void one_cycle (int frames);
public:
static PortInfo port_info [];
void init();
void activate();
void run (int n)

13
plugins/ladspa_effect/caps/HRTF.cc
View File

@@ -1,7 +1,7 @@
/*
HRTF.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -35,9 +35,8 @@
/* //////////////////////////////////////////////////////////////////////// */
void
HRTF::init (double fs)
HRTF::init()
{
normal = NOISE_FLOOR;
h = 0;
}
@@ -74,7 +73,7 @@ HRTF::one_cycle (int frames)
{
d_sample * s = ports[0];
int p = (int) *ports[1];
int p = (int) getport(1);
if (p != pan) set_pan (p);
d_sample * dl = ports[2];
@@ -106,8 +105,6 @@ HRTF::one_cycle (int frames)
F (dl, i, l, adding_gain);
F (dr, i, r, adding_gain);
}
normal = -normal;
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -141,9 +138,9 @@ Descriptor<HRTF>::setup()
Label = "HRTF";
Properties = HARD_RT;
Name = "CAPS: HRTF - Head-related transfer function at elevation 0";
Name = CAPS "HRTF - Head-related transfer function at elevation 0";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

9
plugins/ladspa_effect/caps/HRTF.h
View File

@@ -31,6 +31,7 @@
#include "dsp/util.h"
class HRTF
: public Plugin
{
public:
int pan;
@@ -45,19 +46,13 @@ class HRTF
void set_pan (int p);
d_sample normal;
template <sample_func_t F>
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [4];
d_sample adding_gain;
void init (double _fs);
void init();
void activate()
{
set_pan ((int) *ports[1]);

15
plugins/ladspa_effect/caps/Lorenz.cc
View File

@@ -1,7 +1,7 @@
/*
Lorenz.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -33,9 +33,8 @@
#include "Descriptor.h"
void
Lorenz::init (double _fs)
Lorenz::init()
{
fs = _fs;
lorenz.init (h = .001, 0.1 * frandom());
gain = 0;
}
@@ -47,11 +46,11 @@ Lorenz::one_cycle (int frames)
lorenz.set_rate (*ports[0]);
double g = (gain == *ports[4]) ?
1 : pow (*ports[4] / gain, 1. / (double) frames);
1 : pow (getport(4) / gain, 1. / (double) frames);
d_sample * d = ports[5];
d_sample x, sx = *ports[1], sy = *ports[2], sz = *ports[3];
d_sample x, sx = getport(1), sy = getport(2), sz = getport(3);
for (int i = 0; i < frames; ++i)
{
@@ -63,7 +62,7 @@ Lorenz::one_cycle (int frames)
gain *= g;
}
gain = *ports[4];
gain = getport(4);
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -105,9 +104,9 @@ Descriptor<Lorenz>::setup()
Label = "Lorenz";
Properties = HARD_RT;
Name = "CAPS: Lorenz - The sound of a Lorenz attractor";
Name = CAPS "Lorenz - The sound of a Lorenz attractor";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

13
plugins/ladspa_effect/caps/Lorenz.h
View File

@@ -31,26 +31,21 @@
#include "dsp/Lorenz.h"
class Lorenz
: public Plugin
{
public:
double fs;
d_sample h, gain;
DSP::Lorenz lorenz;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [6];
d_sample adding_gain;
void init (double _fs);
void activate()
{ }
void init();
void activate() {}
void run (int n)
{

4
plugins/ladspa_effect/caps/Makefile.am
View File

@@ -1,7 +1,7 @@
AUTOMAKE_OPTIONS = foreign 1.4
AM_CXXFLAGS := $(AM_CXXFLAGS) -O6 -ffast-math -funroll-loops -ftracer -I../../../include
AM_CXXFLAGS := $(AM_CXXFLAGS) -O2 -I../../../include
DEST = $(DESTDIR)$(libdir)/$(PACKAGE)/ladspa
SOURCES = $(wildcard *.cc)
@@ -15,7 +15,7 @@ clean:
rm -f *.o *.so *.s depend
install: all
strip caps.so
# strip caps.so
install -d $(DEST)
install -m 644 caps.so $(DEST)

21
plugins/ladspa_effect/caps/Pan.cc
View File

@@ -1,7 +1,7 @@
/*
Pan.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -31,11 +31,9 @@
#include "Descriptor.h"
void
Pan::init (double _fs)
Pan::init()
{
fs = _fs;
delay.init ((int) (.040 * fs));
normal = NOISE_FLOOR;
}
void
@@ -64,15 +62,16 @@ Pan::one_cycle (int frames)
d_sample * s = ports[0];
if (pan != *ports[1])
set_pan (*ports[1]);
set_pan (getport(1));
d_sample g = getport(2);
d_sample
width_l = *ports[2] * gain_r,
width_r = *ports[2] * gain_l;
width_l = g * gain_r,
width_r = g * gain_l;
tap.t = (int) (*ports[3] * fs * .001);
tap.t = (int) (getport(3) * fs * .001);
bool mono = *ports[4];
bool mono = getport(4);
d_sample * dl = ports[5];
d_sample * dr = ports[6];
@@ -152,9 +151,9 @@ Descriptor<Pan>::setup()
Label = "Pan";
Properties = HARD_RT;
Name = "CAPS: Pan - Pan and width";
Name = CAPS "Pan - Pan and width";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

11
plugins/ladspa_effect/caps/Pan.h
View File

@@ -50,30 +50,25 @@ class PanTap
};
class Pan
: public Plugin
{
public:
double fs;
d_sample pan;
d_sample gain_l, gain_r;
d_sample normal;
DSP::Delay delay;
PanTap tap;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
inline void set_pan (d_sample);
public:
static PortInfo port_info [];
d_sample * ports [7];
d_sample adding_gain;
void init (double _fs);
void init();
void activate();
void run (int n)

30
plugins/ladspa_effect/caps/Phaser.cc
View File

@@ -1,7 +1,7 @@
/*
Phaser.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -41,18 +41,16 @@ PhaserI::one_cycle (int frames)
if (rate != *ports[1])
{
rate = *ports[1];
rate = getport(1);
lfo.set_f (max (.001, rate * (double) blocksize), fs, lfo.get_phase());
}
double depth = *ports[2];
double spread = 1 + *ports[3];
double fb = *ports[4];
double depth = getport(2);
double spread = 1 + getport(3);
double fb = getport(4);
d_sample * dst = ports[5];
normal = -normal;
while (frames)
{
if (remain == 0) remain = 32;
@@ -126,9 +124,9 @@ Descriptor<PhaserI>::setup()
Label = "PhaserI";
Properties = HARD_RT;
Name = "CAPS: PhaserI - Mono phaser";
Name = CAPS "PhaserI - Mono phaser";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2002-5";
Copyright = "GPL, 2002-7";
/* fill port info and vtable */
autogen();
@@ -142,16 +140,14 @@ PhaserII::one_cycle (int frames)
{
d_sample * s = ports[0];
lorenz.set_rate (*ports[1] * .08);
lorenz.set_rate (getport(1) * .08);
double depth = *ports[2];
double spread = 1 + *ports[3];
double fb = *ports[4];
double depth = getport(2);
double spread = 1 + getport(3);
double fb = getport(4);
d_sample * dst = ports[5];
normal = -normal;
while (frames)
{
if (remain == 0) remain = 32;
@@ -225,9 +221,9 @@ Descriptor<PhaserII>::setup()
Label = "PhaserII";
Properties = HARD_RT;
Name = "CAPS: PhaserII - Mono phaser modulated by a Lorenz fractal";
Name = CAPS "PhaserII - Mono phaser modulated by a Lorenz fractal";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2002-5";
Copyright = "GPL, 2002-7";
/* fill port info and vtable */
autogen();

24
plugins/ladspa_effect/caps/Phaser.h
View File

@@ -58,37 +58,30 @@ class PhaserAP
};
class PhaserI
: public Plugin
{
public:
double fs;
PhaserAP ap[6];
DSP::Sine lfo;
d_sample rate;
d_sample y0;
d_sample normal;
struct {
double bottom, range;
} delay;
template <sample_func_t>
void one_cycle (int frames);
void one_cycle (int frames);
int blocksize, remain;
public:
d_sample * ports [6];
static PortInfo port_info [];
d_sample adding_gain;
void init (double _fs)
void init()
{
fs = _fs;
blocksize = 32;
normal = NOISE_FLOOR;
}
void activate()
@@ -116,6 +109,7 @@ class PhaserI
/* same as above, but filter sweep is controlled by a Lorenz fractal */
class PhaserII
: public Plugin
{
public:
double fs;
@@ -125,28 +119,22 @@ class PhaserII
d_sample rate;
d_sample y0;
d_sample normal;
struct {
double bottom, range;
} delay;
template <sample_func_t>
void one_cycle (int frames);
void one_cycle (int frames);
int blocksize, remain;
public:
d_sample * ports [6];
static PortInfo port_info [];
d_sample adding_gain;
void init (double _fs)
void init()
{
fs = _fs;
blocksize = 32;
normal = NOISE_FLOOR;
lorenz.init();
}

39
plugins/ladspa_effect/caps/Preamp.cc
View File

@@ -1,11 +1,11 @@
/*
Preamp.cc
Copyright 2003-5 Tim Goetze <tim@quitte.de>
Copyright 2003-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
loosely 12AX7-based tube preamp model, 8x oversampling.
Loosely 12AX7-based tube preamp model with 8x oversampling.
*/
/*
This program is free software; you can redistribute it and/or
@@ -31,11 +31,11 @@
#include "Descriptor.h"
void
PreampIII::init (double _fs)
PreampIII::init()
{
this->AmpStub::init (_fs);
this->AmpStub::init();
DSP::RBJ::LoShelve (200 / (_fs), .2, -6, filter.a, filter.b);
DSP::RBJ::LoShelve (200 / fs, .2, -6, filter.a, filter.b);
}
template <sample_func_t F, int OVERSAMPLE>
@@ -43,8 +43,8 @@ void
PreampIII::one_cycle (int frames)
{
d_sample * s = ports[0];
d_sample gain = *ports[1];
d_sample temp = *ports[2] * tube.scale;
d_sample gain = getport(1);
d_sample temp = getport(2) * tube.scale;
d_sample * d = ports[3];
*ports[4] = OVERSAMPLE;
@@ -57,7 +57,7 @@ PreampIII::one_cycle (int frames)
if (g == 0) g = current.g;
/* recursive fade to prevent zipper noise from the 'gain' knob */
double one_over_n = 1. / frames;
double one_over_n = frames > 0 ? 1. / frames : 1;
double gf = pow (current.g / g, one_over_n);
for (int i = 0; i < frames; ++i)
@@ -77,7 +77,6 @@ PreampIII::one_cycle (int frames)
}
current.g = g;
normal = -normal;
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -115,9 +114,9 @@ Descriptor<PreampIII>::setup()
Label = "PreampIII";
Properties = HARD_RT;
Name = "CAPS: PreampIII - Tube preamp emulation";
Name = CAPS "PreampIII - Tube preamp emulation";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2002-5";
Copyright = "GPL, 2002-7";
/* fill port info and vtable */
autogen();
@@ -126,11 +125,10 @@ Descriptor<PreampIII>::setup()
/* //////////////////////////////////////////////////////////////////////// */
void
PreampIV::init (double _fs)
PreampIV::init()
{
this->AmpStub::init (_fs);
tone.init (_fs);
this->AmpStub::init();
tone.init (fs);
}
void
@@ -145,11 +143,11 @@ template <sample_func_t F, int OVERSAMPLE>
void
PreampIV::one_cycle (int frames)
{
double one_over_n = 1. / frames;
double one_over_n = frames > 0 ? 1. / frames : 1;
d_sample * s = ports[0];
d_sample gain = *ports[1];
d_sample temp = *ports[2] * tube.scale;
d_sample gain = getport(1);
d_sample temp = getport(2) * tube.scale;
tone.start_cycle (ports + 3, one_over_n);
@@ -183,7 +181,6 @@ PreampIV::one_cycle (int frames)
}
current.g = g;
normal = -normal;
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -237,9 +234,9 @@ Descriptor<PreampIV>::setup()
Label = "PreampIV";
Properties = HARD_RT;
Name = "CAPS: PreampIV - Tube preamp emulation + tone controls";
Name = CAPS "PreampIV - Tube preamp emulation + tone controls";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2002-5";
Copyright = "GPL, 2002-7";
/* fill port info and vtable */
autogen();

66
plugins/ladspa_effect/caps/README
View File

@@ -1,67 +1,7 @@
This is caps, the C Audio Plugin Suite
This is CAPS, the C Audio Plugin Suite
======================================
Copyright 2002-5 Tim Goetze <tim@quitte.de> and others.
License information
-------------------
See the file "COPYING" for licensing terms & conditions for usage,
and a DISCLAIMER OF ALL WARRANTIES.
What is this caps thing?
------------------------
caps is a collection of refined realtime-capable LADSPA audio DSP
plugins. It helps your computer sound better.
Documentation
-------------
Please point your browser to the file 'caps.html' in this directory.
Web site and latest version
---------------------------
The latest information and package release reside at
For all questions, please be referred to the HTML documentation of
this software package in the file 'caps.html', also available from
http://quitte.de/dsp/caps.html
Installation
============
At the shell prompt, enter (without the '$' denoting the prompt itself):
$ make
then, as the superuser, enter:
# make install
and you're done. You can now use the plugins in any LADSPA-aware audio
application.
Please note that the default installation path is /usr/local/lib/ladspa;
if you prefer another place, you need to edit the makefile, or install
manually.
Bug reports
-----------
Bug reports should be sent to <tim@quitte.de>. Please include a detailed
description of the misbehaviour and instructions on how to reproduce the
bug.
Troubleshooting
---------------
caps is known to compile well with gnu make, gcc 2.95, 3.3 and 4.0.
If you have trouble compiling it, please send mail containing the
output of the 'make' command to <tim@quitte.de>.

51
plugins/ladspa_effect/caps/Reverb.cc
View File

@@ -1,7 +1,7 @@
/*
Reverb.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -57,11 +57,8 @@ JVRev::default_length[9] = {
};
void
JVRev::init (double _fs)
JVRev::init()
{
fs = _fs;
normal = NOISE_FLOOR;
memcpy (length, default_length, sizeof (length));
if (fs != 44100)
@@ -114,7 +111,7 @@ JVRev::activate()
left.reset();
right.reset();
set_t60 (*ports[1]);
set_t60 (getport(1));
}
template <sample_func_t F>
@@ -124,15 +121,13 @@ JVRev::one_cycle (int frames)
d_sample * s = ports[0];
if (t60 != *ports[1])
set_t60 (*ports[1]);
set_t60 (getport(1));
double wet = *ports[2], dry = 1 - wet;
double wet = getport(2), dry = 1 - wet;
d_sample * dl = ports[3];
d_sample * dr = ports[4];
normal = -normal;
for (int i = 0; i < frames; ++i)
{
d_sample x = s[i], a = x + normal;
@@ -191,9 +186,9 @@ Descriptor<JVRev>::setup()
Label = "JVRev";
Properties = HARD_RT;
Name = "CAPS: JVRev - Stanford-style reverb from STK";
Name = CAPS "JVRev - Stanford-style reverb from STK";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
@@ -202,9 +197,8 @@ Descriptor<JVRev>::setup()
/* //////////////////////////////////////////////////////////////////////// */
void
PlateStub::init (double _fs)
PlateStub::init()
{
fs = _fs;
f_lfo = -1;
# define L(i) ((int) (l[i] * fs))
@@ -257,8 +251,6 @@ PlateStub::init (double _fs)
dediff1 = .723;
dediff2 = .729;
normal = NOISE_FLOOR;
}
inline void
@@ -321,15 +313,15 @@ Plate::one_cycle (int frames)
{
d_sample * s = ports[0];
input.bandwidth.set (exp (-M_PI * (1. - *ports[1])));
input.bandwidth.set (exp (-M_PI * (1. - getport(1))));
d_sample decay = *ports[2];
d_sample decay = getport(2);
double damp = exp (-M_PI * *ports[3]);
double damp = exp (-M_PI * getport(3));
tank.damping[0].set (damp);
tank.damping[1].set (damp);
d_sample blend = *ports[4], dry = 1 - blend;
d_sample blend = getport(4), dry = 1 - blend;
d_sample * dl = ports[5];
d_sample * dr = ports[6];
@@ -339,8 +331,8 @@ Plate::one_cycle (int frames)
for (int i = 0; i < frames; ++i)
{
d_sample x = s[i] + normal;
normal = -normal;
d_sample x = s[i] + normal;
d_sample xl, xr;
@@ -396,9 +388,9 @@ Descriptor<Plate>::setup()
Label = "Plate";
Properties = HARD_RT;
Name = "CAPS: Plate - Versatile plate reverb";
Name = CAPS "Plate - Versatile plate reverb";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
@@ -413,15 +405,15 @@ Plate2x2::one_cycle (int frames)
d_sample * sl = ports[0];
d_sample * sr = ports[1];
input.bandwidth.set (exp (-M_PI * (1. - *ports[2])));
input.bandwidth.set (exp (-M_PI * (1. - getport(2))));
d_sample decay = *ports[3];
d_sample decay = getport(3);
double damp = exp (-M_PI * *ports[4]);
double damp = exp (-M_PI * getport(4));
tank.damping[0].set (damp);
tank.damping[1].set (damp);
d_sample blend = *ports[5], dry = 1 - blend;
d_sample blend = getport(5), dry = 1 - blend;
d_sample * dl = ports[6];
d_sample * dr = ports[7];
@@ -432,7 +424,6 @@ Plate2x2::one_cycle (int frames)
for (int i = 0; i < frames; ++i)
{
normal = -normal;
d_sample x = (sl[i] + sr[i] + normal) * .5;
d_sample xl, xr;
@@ -493,9 +484,9 @@ Descriptor<Plate2x2>::setup()
Label = "Plate2x2";
Properties = HARD_RT;
Name = "CAPS: Plate2x2 - Versatile plate reverb, stereo inputs";
Name = CAPS "Plate2x2 - Versatile plate reverb, stereo inputs";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

21
plugins/ladspa_effect/caps/Reverb.h
View File

@@ -86,10 +86,10 @@ class JVComb
};
class JVRev
: public Plugin
{
public:
static int default_length[9];
double fs;
d_sample t60;
Lattice allpass [3];
@@ -99,8 +99,6 @@ class JVRev
double apc;
d_sample normal;
template <sample_func_t F>
void one_cycle (int frames);
@@ -110,12 +108,8 @@ class JVRev
public:
static PortInfo port_info [];
d_sample * ports [5];
d_sample adding_gain;
void init (double _fs);
void init();
void activate();
void run (int n)
@@ -165,9 +159,9 @@ class ModLattice
};
class PlateStub
: public Plugin
{
public:
double fs;
d_sample f_lfo;
d_sample indiff1, indiff2, dediff1, dediff2;
@@ -185,13 +179,8 @@ class PlateStub
int taps[12];
} tank;
d_sample normal;
public:
d_sample adding_gain;
void init (double _fs);
void init();
void activate()
{
input.bandwidth.reset();
@@ -228,7 +217,6 @@ class Plate
public:
static PortInfo port_info [];
d_sample * ports [7];
void run (int n)
{
@@ -252,7 +240,6 @@ class Plate2x2
public:
static PortInfo port_info [];
d_sample * ports [8];
void run (int n)
{

23
plugins/ladspa_effect/caps/Roessler.cc
View File

@@ -1,7 +1,7 @@
/*
Roessler.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -33,9 +33,8 @@
#include "Descriptor.h"
void
Roessler::init (double _fs)
Roessler::init()
{
fs = _fs;
roessler.init (h = .001, frandom());
gain = 0;
}
@@ -44,17 +43,17 @@ template <sample_func_t F>
void
Roessler::one_cycle (int frames)
{
roessler.set_rate (*ports[0]);
roessler.set_rate (getport(0));
double g = (gain == *ports[4]) ?
1 : pow (*ports[4] / gain, 1. / (double) frames);
double g = (gain == getport(4)) ?
1 : pow (getport(4) / gain, 1. / (double) frames);
d_sample * d = ports[5];
d_sample x,
sx = .043 * *ports[1],
sy = .051 * *ports[2],
sz = .018 * *ports[3];
sx = .043 * getport(1),
sy = .051 * getport(2),
sz = .018 * getport(3);
for (int i = 0; i < frames; ++i)
{
@@ -69,7 +68,7 @@ Roessler::one_cycle (int frames)
gain *= g;
}
gain = *ports[4];
gain = getport(4);
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -111,9 +110,9 @@ Descriptor<Roessler>::setup()
Label = "Roessler";
Properties = HARD_RT;
Name = "CAPS: Roessler - The sound of a Roessler attractor";
Name = CAPS "Roessler - The sound of a Roessler attractor";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

9
plugins/ladspa_effect/caps/Roessler.h
View File

@@ -31,9 +31,9 @@
#include "dsp/Roessler.h"
class Roessler
: public Plugin
{
public:
double fs;
d_sample h, gain;
DSP::Roessler roessler;
@@ -43,14 +43,11 @@ class Roessler
public:
static PortInfo port_info [];
d_sample * ports [6];
d_sample adding_gain;
void init (double _fs);
void activate()
{ }
void init();
void activate() {}
void run (int n)
{

22
plugins/ladspa_effect/caps/Scape.cc
View File

@@ -1,7 +1,7 @@
/*
Scape.cc
Copyright 2004-6 Tim Goetze <tim@quitte.de>
Copyright 2004-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -66,15 +66,15 @@ Scape::one_cycle (int frames)
// double one_over_n = 1 / (double) frames;
/* delay times */
double t1 = fs * 60. / *ports[1];
int div = (int) *ports[2];
double t1 = fs * 60. / getport(1);
int div = (int) getport(2);
double t2 = t1 * dividers[div];
fb = *ports[3];
fb = getport(3);
double dry = *ports[4];
double dry = getport(4);
dry = dry * dry;
double blend = *ports[5];
double blend = getport(5);
d_sample * dl = ports[6];
d_sample * dr = ports[7];
@@ -92,7 +92,6 @@ Scape::one_cycle (int frames)
period = t2 * .5;
float f, q;
//fprintf (stderr, "%.3f: %d\n", period, (int) period);
f = frandom2();
svf[0].set_f_Q (300 + 300 * f / fs, .3);
svf[3].set_f_Q (300 + 600 * 2 * f / fs, .6);
@@ -107,10 +106,13 @@ Scape::one_cycle (int frames)
int n = min ((int) period, frames);
if (n < 1)
{
/* not reached */
#ifdef DEBUG
fprintf (stderr, "Scape: %d - %d/%d frames, t2 = %.3f?!?\n", (int) period, n, frames, t2);
#endif
return;
}
/* sample loop */
for (int i = 0; i < n; ++i)
{
@@ -193,9 +195,9 @@ Descriptor<Scape>::setup()
Label = "Scape";
Properties = HARD_RT;
Name = "CAPS: Scape - Stereo delay + Filters";
Name = CAPS "Scape - Stereo delay + Filters";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-6";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

21
plugins/ladspa_effect/caps/Scape.h
View File

@@ -38,12 +38,11 @@
typedef DSP::SVF<1> SVF;
class Scape
: public Plugin
{
public:
double fs;
d_sample time, fb;
double period;
d_sample normal; /* denormal protection */
DSP::Lorenz lfo[2];
DSP::Delay delay;
@@ -51,28 +50,14 @@ class Scape
DSP::OnePoleHP hipass[4];
template <sample_func_t>
void one_cycle (int frames);
void one_cycle (int frames);
public:
d_sample * ports [8];
/*
* in
* bpm
* div
* feedback
* dry
* blend
* out:l
* out:r
*/
static PortInfo port_info [];
d_sample adding_gain;
void init (double _fs)
void init()
{
fs = _fs;
delay.init ((int) (2.01 * fs)); /* two seconds = 30 bpm + */
normal = NOISE_FLOOR;
for (int i = 0; i < 2; ++i)
lfo[i].init(),
lfo[i].set_rate (.00000001 * fs);

17
plugins/ladspa_effect/caps/Sin.cc
View File

@@ -1,7 +1,7 @@
/*
Sin.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -31,9 +31,8 @@
#include "Descriptor.h"
void
Sin::init (double _fs)
Sin::init()
{
fs = _fs;
sin.set_f (f = .005, fs, 0);
gain = 0;
}
@@ -43,10 +42,10 @@ void
Sin::one_cycle (int frames)
{
if (f != *ports[0])
sin.set_f (f = *ports[0], fs, sin.get_phase());
sin.set_f (f = getport(0), fs, sin.get_phase());
double g = (gain == *ports[1]) ?
1 : pow (*ports[1] / gain, 1. / (double) frames);
1 : pow (getport(1) / gain, 1. / (double) frames);
d_sample * d = ports[2];
@@ -56,7 +55,7 @@ Sin::one_cycle (int frames)
gain *= g;
}
gain = *ports[1];
gain = getport(1);
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -67,7 +66,7 @@ Sin::port_info [] =
{
"f",
INPUT | CONTROL,
{BOUNDED | FS | LOG | DEFAULT_440, 0, .5}
{BOUNDED | LOG | DEFAULT_100, 0.0001, 20000}
}, {
"volume",
INPUT | CONTROL,
@@ -86,9 +85,9 @@ Descriptor<Sin>::setup()
Label = "Sin";
Properties = HARD_RT;
Name = "CAPS: Sin - Sine wave generator";
Name = CAPS "Sin - Sine wave generator";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

13
plugins/ladspa_effect/caps/Sin.h
View File

@@ -31,26 +31,21 @@
#include "dsp/Sine.h"
class Sin
: public Plugin
{
public:
double fs;
d_sample f, gain;
DSP::Sine sin;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [3];
d_sample adding_gain;
void init (double _fs);
void activate()
{ }
void init();
void activate() {}
void run (int n)
{

215
plugins/ladspa_effect/caps/SweepVF.cc
View File

@@ -1,7 +1,7 @@
/*
SweepVF.cc
Copyright 2002-5 Tim Goetze <tim@quitte.de>
Copyright 2002-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -33,21 +33,21 @@
#include "SweepVF.h"
#include "Descriptor.h"
#include "dsp/RBJ.h"
void
SweepVFI::init (double _fs)
SweepVFI::init()
{
fs = _fs;
f = .1;
Q = .1;
lorenz.init();
normal = NOISE_FLOOR;
}
void
SweepVFI::activate()
{
svf.reset();
svf.set_f_Q (f = *ports[1] / fs, Q = *ports[2]);
svf.set_f_Q (f = getport(1) / fs, Q = getport(2));
}
template <sample_func_t F>
@@ -62,12 +62,12 @@ SweepVFI::one_cycle (int frames)
double one_over_blocks = 1 / (double) blocks;
/* cheesy linear interpolation for f, works well though. */
double df = (*ports[1] / fs - f) * one_over_blocks;
double dQ = (*ports[2] - Q) * one_over_blocks;
double df = (getport(1) / fs - f) * one_over_blocks;
double dQ = (getport(2) - Q) * one_over_blocks;
svf.set_out ((int) *ports[3]);
svf.set_out ((int) getport(3));
lorenz.set_rate (*ports[7]);
lorenz.set_rate (getport(7));
d_sample * d = ports[8];
@@ -76,11 +76,11 @@ SweepVFI::one_cycle (int frames)
lorenz.step();
double modulation =
*ports[4] * lorenz.get_x() +
*ports[5] * lorenz.get_y() +
*ports[6] * lorenz.get_z();
getport(4) * lorenz.get_x() +
getport(5) * lorenz.get_y() +
getport(6) * lorenz.get_z();
double scale = *ports[4] + *ports[5] + *ports[6];
double scale = getport(4) + getport(5) + getport(6);
modulation *= scale * f;
svf.set_f_Q (max (.001, f + modulation), Q);
@@ -98,10 +98,8 @@ SweepVFI::one_cycle (int frames)
Q += dQ;
}
normal = -normal;
f = *ports[1] / fs;
Q = *ports[2];
f = getport(1) / fs;
Q = getport(2);
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -116,7 +114,7 @@ SweepVFI::port_info [] =
}, {
"f",
INPUT | CONTROL,
{BOUNDED | FS | DEFAULT_LOW, 0.002, 0.08}
{BOUNDED | LOG | DEFAULT_LOW, 83, 3383}
}, {
"Q",
INPUT | CONTROL,
@@ -155,9 +153,9 @@ Descriptor<SweepVFI>::setup()
Label = "SweepVFI";
Properties = HARD_RT;
Name = "CAPS: SweepVFI - Resonant filter, f swept by a Lorenz fractal";
Name = CAPS "SweepVFI - Resonant filter swept by a Lorenz fractal";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
@@ -166,21 +164,19 @@ Descriptor<SweepVFI>::setup()
/* //////////////////////////////////////////////////////////////////////// */
void
SweepVFII::init (double _fs)
SweepVFII::init()
{
fs = _fs;
f = .1;
Q = .1;
lorenz1.init();
lorenz2.init();
normal = NOISE_FLOOR;
}
void
SweepVFII::activate()
{
svf.reset();
svf.set_f_Q (f = *ports[1] / fs, Q = *ports[2]);
svf.set_f_Q (f = getport(1) / fs, Q = getport(2));
}
template <sample_func_t F>
@@ -195,13 +191,13 @@ SweepVFII::one_cycle (int frames)
double one_over_blocks = 1 / (double) blocks;
/* cheesy linear interpolation for f, works well though. */
double df = (*ports[1] / fs - f) * one_over_blocks;
double dQ = (*ports[2] - Q) * one_over_blocks;
double df = (getport(1) / fs - f) * one_over_blocks;
double dQ = (getport(2) - Q) * one_over_blocks;
svf.set_out ((int) *ports[3]);
svf.set_out ((int) getport(3));
lorenz1.set_rate (*ports[7]);
lorenz2.set_rate (*ports[11]);
lorenz1.set_rate (getport(7));
lorenz2.set_rate (getport(11));
d_sample * d = ports[12];
@@ -211,11 +207,11 @@ SweepVFII::one_cycle (int frames)
lorenz1.step();
double modulation1 =
*ports[4] * lorenz1.get_x() +
*ports[5] * lorenz1.get_y() +
*ports[6] * lorenz1.get_z();
getport(4) * lorenz1.get_x() +
getport(5) * lorenz1.get_y() +
getport(6) * lorenz1.get_z();
double scale1 = *ports[4] + *ports[5] + *ports[6];
double scale1 = getport(4) + getport(5) + getport(6);
modulation1 *= scale1 * f;
@@ -223,11 +219,11 @@ SweepVFII::one_cycle (int frames)
lorenz2.step();
double modulation2 =
*ports[8] * lorenz2.get_x() +
*ports[9] * lorenz2.get_y() +
*ports[10] * lorenz2.get_z();
getport(8) * lorenz2.get_x() +
getport(9) * lorenz2.get_y() +
getport(10) * lorenz2.get_z();
double scale2 = *ports[8] + *ports[9] + *ports[10];
double scale2 = getport(8) + getport(9) + getport(10);
/* enforce Q limit */
double q = Q + (modulation2 * scale2 * Q);
@@ -248,10 +244,8 @@ SweepVFII::one_cycle (int frames)
Q += dQ;
}
normal = -normal;
f = *ports[1] / fs;
Q = *ports[2];
f = getport(1) / fs;
Q = getport(2);
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -266,7 +260,7 @@ SweepVFII::port_info [] =
}, {
"f",
INPUT | CONTROL,
{BOUNDED | FS | DEFAULT_LOW, 0.002, 0.08}
{BOUNDED | LOG | DEFAULT_LOW, 83, 3383}
}, {
"Q",
INPUT | CONTROL,
@@ -321,11 +315,144 @@ Descriptor<SweepVFII>::setup()
Label = "SweepVFII";
Properties = HARD_RT;
Name = "CAPS: SweepVFII - Resonant filter, f and Q swept by a Lorenz fractal";
Name = CAPS "SweepVFII - Resonant filter, f and Q swept by a Lorenz fractal";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
}
/* //////////////////////////////////////////////////////////////////////// */
void
AutoWah::init()
{
f = 800 / fs;
Q = .5;
}
void
AutoWah::activate()
{
svf.reset();
svf.set_f_Q (f = getport(1) / fs, Q = getport(2));
svf.set_out (DSP::SVF<1>::Band);
/* hi-passing input for envelope RMS calculation */
hp.set_f (250. / fs);
/* smoothing the envelope at 20 Hz */
DSP::RBJ::LP (20. * BLOCK_SIZE / fs, .6, filter.a, filter.b);
rms.reset();
hp.reset();
filter.reset();
}
template <sample_func_t F>
void
AutoWah::one_cycle (int frames)
{
d_sample * s = ports[0];
int blocks = frames / BLOCK_SIZE;
if (frames & (BLOCK_SIZE - 1))
++blocks;
double one_over_blocks = 1 / (double) blocks;
/* cheesy linear interpolation for f, works well though. */
double df = (getport(1) / fs - f) * one_over_blocks;
double dQ = (getport(2) - Q) * one_over_blocks;
double scale = getport(3);
d_sample * d = ports[4];
while (frames)
{
double m = rms.rms();
m = filter.process (m + normal);
/* Leaving debug output in your code is cheesy! */
/*
static int _turn = 0;
if (_turn++ % 100 == 0)
fprintf (stderr, "%.4f\n", m);
*/
m *= scale * .08;
svf.set_f_Q (max (.001, f + m), Q);
int n = min (frames, BLOCK_SIZE);
for (int i = 0; i < n; ++i)
{
d_sample x = s[i] + normal;
/* A stacked SVF in bandpass mode is rather quiet, which is
* compensated here */
F (d, i, 2 * svf.process (x), adding_gain);
/* for envelope calculation, prefer high f content */
x = hp.process (x);
rms.store (x * x);
}
s += n;
d += n;
frames -= n;
f += df;
Q += dQ;
normal = -normal;
}
f = getport(1) / fs;
Q = getport(2);
}
/* //////////////////////////////////////////////////////////////////////// */
PortInfo
AutoWah::port_info [] =
{
{
"in",
INPUT | AUDIO,
{0, 0, 0}
}, {
"f",
INPUT | CONTROL,
{BOUNDED | LOG | DEFAULT_LOW, 43, 933}
}, {
"Q",
INPUT | CONTROL,
{BOUNDED | DEFAULT_LOW, 0.001, .999}
}, {
"depth",
INPUT | CONTROL,
{BOUNDED | DEFAULT_MID, 0, 1}
}, {
"out",
OUTPUT | AUDIO,
{0}
}
};
template <> void
Descriptor<AutoWah>::setup()
{
UniqueID = 2593;
Label = "AutoWah";
Properties = HARD_RT;
Name = CAPS "AutoWah - Resonant envelope-following filter";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
}

73
plugins/ladspa_effect/caps/SweepVF.h
View File

@@ -1,7 +1,7 @@
/*
SweepVF.h
Copyright 2004-5 Tim Goetze <tim@quitte.de>
Copyright 2004-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -10,6 +10,8 @@
SweepVFII, the same with Q being modulated by a second fractal.
AutoWah, SVF being modulated by 'instant' amplitude (envelope).
*/
/*
This program is free software; you can redistribute it and/or
@@ -32,10 +34,16 @@
#define _SWEEP_VF_H_
#include "dsp/SVF.h"
#include "dsp/Lorenz.h"
#include "dsp/Roessler.h"
#include "dsp/RMS.h"
#include "dsp/BiQuad.h"
#include "dsp/OnePole.h"
class SweepVFI
: public Plugin
{
public:
double fs;
@@ -51,19 +59,13 @@ class SweepVFI
DSP::StackedSVF<1,2> svf;
DSP::Lorenz lorenz;
d_sample normal;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [9];
d_sample adding_gain;
void init (double _fs);
void init();
void activate();
void run (int n)
@@ -78,6 +80,45 @@ class SweepVFI
};
class SweepVFII
: public Plugin
{
public:
/* svf parameters */
d_sample f, Q;
/* needs to be a power of two */
enum {
BLOCK_SIZE = 32
};
DSP::StackedSVF<1,2> svf;
DSP::Lorenz lorenz1;
DSP::Lorenz lorenz2;
template <sample_func_t F>
void one_cycle (int frames);
public:
static PortInfo port_info [];
void init();
void activate();
void run (int n)
{
one_cycle<store_func> (n);
}
void run_adding (int n)
{
one_cycle<adding_func> (n);
}
};
/* //////////////////////////////////////////////////////////////////////// */
class AutoWah
: public Plugin
{
public:
double fs;
@@ -91,22 +132,18 @@ class SweepVFII
};
DSP::StackedSVF<1,2> svf;
DSP::Lorenz lorenz1;
DSP::Lorenz lorenz2;
DSP::RMS rms;
d_sample normal;
DSP::BiQuad filter;
DSP::OnePoleHP hp;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [13];
d_sample adding_gain;
void init (double _fs);
void init();
void activate();
void run (int n)

2
plugins/ladspa_effect/caps/ToneControls.cc
View File

@@ -1,7 +1,7 @@
/*
ToneControls.cc
Copyright 2004-5 Tim Goetze <tim@quitte.de>
Copyright 2004-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/

195
plugins/ladspa_effect/caps/ToneStack.cc Normal file
View File

@@ -0,0 +1,195 @@
/*
ToneStack.cc
Copyright 2006-7
David Yeh <dtyeh@ccrma.stanford.edu>
Tim Goetze <tim@quitte.de> (cosmetics)
Tone Stack emulation.
*
*/
/*
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
#include "basics.h"
#include "ToneStack.h"
#include "Descriptor.h"
#include "dsp/tonestack/ks_tab.h"
#include "dsp/tonestack/vs_tab.h"
DSP::TSParameters
DSP::ToneStack::presets[] = {
/* for convenience, temporarily define k and MOhms as well as nF and pF */
#define k * 1000
#define M * 1000000
#define nF * 1e-9
#define pF * 1e-12
/* parameter order is R1 - R4, C1 - C3 */
/* { 250000, 1000000, 25000, 56000, 0.25e-9, 20e-9, 20e-9 }, DY */
/* Fender */
{250 k, 1 M, 25 k, 56 k, 250 pF, 20 nF, 20 nF}, /* 59 Bassman 5F6-A */
{250 k, 250 k, 10 k, 100 k, 120 pF, 100 nF, 47 nF}, /* 69 Twin Reverb AA270 */
{250 k, 250 k, 4.8 k, 100 k, 250 pF, 100 nF, 47 nF}, /* 64 Princeton AA1164 */
/* Marshall */
{220 k, 1 M, 22 k, 33 k, 470 pF, 22 nF, 22 nF}, /* 59/81 JCM-800 Lead 100 2203 */
/* R4 is a 10 k fixed + 100 k pot in series actually */
{250 k, 1 M, 25 k, 56 k, 500 pF, 22 nF, 22 nF}, /* 81 2000 Lead */
#if 0
{220 k, 1 M, 22 k, 33 k, 470 pF, 22 nF, 22 nF}, /* 90 JCM-900 Master 2100 (same as JCM-800) */
{250 k, 1 M, 25 k, 33 k, 500 pF, 22 nF, 22 nF}, /* 67 Major Lead 200 */
{250 k, 250 k, 25 k, 56 k, 250 pF, 47 nF, 47 nF}, /* undated M2199 30W solid state */
#endif
/* Vox -- R3 is fixed (circuit differs anyway) */
{1 M, 1 M, 10 k, 100 k, 50 pF, 22 nF, 22 nF}, /* 59/86 AC-30 */
#undef k
#undef M
#undef nF
#undef pF
};
int DSP::ToneStack::n_presets = TS_N_PRESETS;
void
ToneStack::activate()
{
tonestack.activate (ports + 2);
}
template <sample_func_t F>
void
ToneStack::one_cycle (int frames)
{
d_sample * s = ports[0];
tonestack.start_cycle (ports + 1);
d_sample * d = ports[5];
for (int i = 0; i < frames; ++i)
{
register d_sample a = s[i];
a = tonestack.process (a + normal);
F (d, i, a, adding_gain);
}
}
PortInfo
ToneStack::port_info [] =
{
{
"in",
INPUT | AUDIO,
{BOUNDED, -1, 1}
}, {
"model",
INPUT | CONTROL,
{BOUNDED | DEFAULT_0 | INTEGER, 0, TS_N_PRESETS - 1}
}, {
"bass",
INPUT | CONTROL,
{BOUNDED | DEFAULT_MID, 0, 1}
}, {
"mid",
INPUT | CONTROL,
{BOUNDED | DEFAULT_MID, 0, 1}
}, {
"treble",
INPUT | CONTROL,
{BOUNDED | DEFAULT_MID, 0, 1}
}, {
"out",
OUTPUT | AUDIO,
{0}
}
};
template <> void
Descriptor<ToneStack>::setup()
{
UniqueID = 2589;
Label = "ToneStack";
Properties = HARD_RT;
Name = CAPS "ToneStack - Tone stack emulation";
Maker = "David Yeh <dtyeh@ccrma.stanford.edu>";
Copyright = "GPL, 2006-7";
/* fill port info and vtable */
autogen();
}
/* //////////////////////////////////////////////////////////////////////// */
template <sample_func_t F>
void
ToneStackLT::one_cycle (int frames)
{
d_sample * s = ports[0];
tonestack.updatecoefs (ports + 1);
d_sample * d = ports[4];
for (int i = 0; i < frames; ++i)
{
register d_sample a = s[i];
a = tonestack.process (a + normal);
F (d, i, a, adding_gain);
}
}
PortInfo
ToneStackLT::port_info [] =
{
{
"in",
INPUT | AUDIO,
{BOUNDED, -1, 1}
}, {
"bass",
INPUT | CONTROL,
{BOUNDED | DEFAULT_MID, 0, 1}
}, {
"mid",
INPUT | CONTROL,
{BOUNDED | DEFAULT_MID, 0, 1}
}, {
"treble",
INPUT | CONTROL,
{BOUNDED | DEFAULT_MID, 0, 1}
}, {
"out",
OUTPUT | AUDIO,
{0}
}
};
template <> void
Descriptor<ToneStackLT>::setup()
{
UniqueID = 2590;
Label = "ToneStackLT";
Properties = HARD_RT;
Name = CAPS "ToneStackLT - Tone stack emulation, lattice filter 44.1";
Maker = "David Yeh <dtyeh@ccrma.stanford.edu>";
Copyright = "GPL, 2006-7";
/* fill port info and vtable */
autogen();
}

98
plugins/ladspa_effect/caps/ToneStack.h Normal file
View File

@@ -0,0 +1,98 @@
/*
ToneStack.h
Copyright 2006-7
David Yeh <dtyeh@ccrma.stanford.edu>
Tim Goetze <tim@quitte.de> (cosmetics)
Tone Stack emulation.
*/
/*
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
#ifndef _TONESTACK_H_
#define _TONESTACK_H_
#include "dsp/util.h"
#include "dsp/windows.h"
#include "dsp/ToneStack.h"
class ToneStack
: public Plugin
{
private:
DSP::ToneStack tonestack;
template <sample_func_t F>
void one_cycle (int frames);
public:
static PortInfo port_info [];
void init()
{
tonestack.init (fs);
}
void activate();
void run (int n)
{
one_cycle<store_func> (n);
}
void run_adding (int n)
{
one_cycle<adding_func> (n);
}
};
/* /////////////////////////////////////////////////////////////////////// */
class ToneStackLT
: public Plugin
{
private:
DSP::ToneStackLT tonestack;
template <sample_func_t F>
void one_cycle (int frames);
public:
static PortInfo port_info [];
void init()
{
tonestack.init (fs);
}
void activate()
{ tonestack.activate (ports + 1); }
void run (int n)
{
one_cycle<store_func> (n);
}
void run_adding (int n)
{
one_cycle<adding_func> (n);
}
};
#endif /* _TONESTACK_H_ */

52
plugins/ladspa_effect/caps/VCO.cc
View File

@@ -1,7 +1,7 @@
/*
VCO.cc
Copyright 2004-5 Tim Goetze <tim@quitte.de>
Copyright 2004-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -32,10 +32,8 @@
#include "Descriptor.h"
void
VCOs::init (double _fs)
VCOs::init()
{
fs = _fs;
/* going a fair bit lower than nominal with fc because the filter
* rolloff is not as steep as we might like it to be. */
double f = .5 * M_PI / OVERSAMPLE;
@@ -59,11 +57,11 @@ template <sample_func_t F>
void
VCOs::one_cycle (int frames)
{
vco.set_f (*ports[0], OVERSAMPLE * fs);
vco.set_saw_square (*ports[1], *ports[2]);
vco.set_f (getport(0), OVERSAMPLE * fs);
vco.set_saw_square (getport(1), getport(2));
double g = (gain == *ports[3]) ?
1 : pow (*ports[3] / gain, 1. / (double) frames);
1 : pow (getport(3) / gain, 1. / (double) frames);
d_sample * d = ports[4];
@@ -77,7 +75,7 @@ VCOs::one_cycle (int frames)
gain *= g;
}
gain = *ports[3];
gain = getport(3);
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -88,7 +86,7 @@ VCOs::port_info [] =
{
"f",
INPUT | CONTROL,
{BOUNDED | FS | DEFAULT_440, 0, .1}
{BOUNDED | LOG | DEFAULT_100, 1, 5751}
}, {
"tri .. saw",
INPUT | CONTROL,
@@ -105,10 +103,6 @@ VCOs::port_info [] =
"out",
OUTPUT | AUDIO,
{0}
}, {
"latency",
OUTPUT | CONTROL,
{0}
}
};
@@ -119,9 +113,9 @@ Descriptor<VCOs>::setup()
Label = "VCOs";
Properties = HARD_RT;
Name = "CAPS: VCOs - Virtual 'analogue' oscillator";
Name = CAPS "VCOs - Virtual 'analogue' oscillator";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();
@@ -130,10 +124,8 @@ Descriptor<VCOs>::setup()
/* //////////////////////////////////////////////////////////////////////// */
void
VCOd::init (double _fs)
VCOd::init()
{
fs = _fs;
/* going a fair bit lower than nominal with fc because the filter
* rolloff is not as steep as we might like it to be. */
double f = .5 * M_PI / OVERSAMPLE;
@@ -157,16 +149,16 @@ template <sample_func_t F>
void
VCOd::one_cycle (int frames)
{
vco.set_f (*ports[0], OVERSAMPLE * fs, *ports[5]);
vco.set_f (getport(0), OVERSAMPLE * fs, getport(5));
vco.vco[0].set_saw_square (*ports[1], *ports[2]);
vco.vco[1].set_saw_square (*ports[3], *ports[4]);
vco.vco[0].set_saw_square (getport(1), getport(2));
vco.vco[1].set_saw_square (getport(3), getport(4));
vco.set_sync (*ports[6]);
vco.set_blend (*ports[7]);
vco.set_sync (getport(6));
vco.set_blend (getport(7));
double g = (gain == *ports[8]) ?
1 : pow (*ports[8] / gain, 1. / (double) frames);
1 : pow (getport(8) / gain, 1. / (double) frames);
d_sample * d = ports[9];
@@ -180,7 +172,7 @@ VCOd::one_cycle (int frames)
gain *= g;
}
gain = *ports[8];
gain = getport(8);
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -191,7 +183,7 @@ VCOd::port_info [] =
{
"f",
INPUT | CONTROL,
{BOUNDED | FS | DEFAULT_440, 0, .1}
{BOUNDED | LOG | DEFAULT_100, 1, 5751}
}, {
"1: tri .. saw",
INPUT | CONTROL,
@@ -228,10 +220,6 @@ VCOd::port_info [] =
"out",
OUTPUT | AUDIO,
{0}
}, {
"latency",
OUTPUT | CONTROL,
{0}
}
};
@@ -242,9 +230,9 @@ Descriptor<VCOd>::setup()
Label = "VCOd";
Properties = HARD_RT;
Name = "CAPS: VCOd - Double VCO with detune and hard sync options";
Name = CAPS "VCOd - Double VCO with detune and hard sync options";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

21
plugins/ladspa_effect/caps/VCO.h
View File

@@ -37,9 +37,9 @@
#include "dsp/windows.h"
class VCOs
: public Plugin
{
public:
double fs;
d_sample f, gain;
/* ok to just change these as you please, 4/32 works ok, sortof. */
@@ -54,27 +54,21 @@ class VCOs
DSP::FIR down;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info[];
d_sample * ports [6];
d_sample adding_gain;
VCOs()
: down (FIR_SIZE)
{ }
void init (double _fs);
void init();
void activate()
{
gain = *ports[3];
down.reset();
vco.reset();
/* latency */
*ports[5] = OVERSAMPLE;
}
void run (int n)
@@ -91,6 +85,7 @@ class VCOs
/* //////////////////////////////////////////////////////////////////////// */
class VCOd
: public Plugin
{
public:
double fs;
@@ -112,23 +107,17 @@ class VCOd
public:
static PortInfo port_info[];
d_sample * ports [11];
d_sample adding_gain;
VCOd()
: down (FIR_SIZE)
{ }
void init (double _fs);
void init();
void activate()
{
gain = *ports[8];
down.reset();
vco.reset();
/* latency */
*ports[10] = OVERSAMPLE;
}
void run (int n)

10
plugins/ladspa_effect/caps/White.cc
View File

@@ -1,7 +1,7 @@
/*
White.cc
Copyright 2004-5 Tim Goetze <tim@quitte.de>
Copyright 2004-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
@@ -35,7 +35,7 @@ void
White::one_cycle (int frames)
{
double g = (gain == *ports[0]) ?
1 : pow (*ports[0] / gain, 1. / (double) frames);
1 : pow (getport(0) / gain, 1. / (double) frames);
d_sample * d = ports[1];
@@ -45,7 +45,7 @@ White::one_cycle (int frames)
gain *= g;
}
gain = *ports[0];
gain = getport(0);
}
/* //////////////////////////////////////////////////////////////////////// */
@@ -71,9 +71,9 @@ Descriptor<White>::setup()
Label = "White";
Properties = HARD_RT;
Name = "CAPS: White - White noise generator";
Name = CAPS "White - White noise generator";
Maker = "Tim Goetze <tim@quitte.de>";
Copyright = "GPL, 2004-5";
Copyright = "GPL, 2004-7";
/* fill port info and vtable */
autogen();

10
plugins/ladspa_effect/caps/White.h
View File

@@ -31,6 +31,7 @@
#include "dsp/White.h"
class White
: public Plugin
{
public:
d_sample gain;
@@ -38,17 +39,12 @@ class White
DSP::White white;
template <sample_func_t F>
void one_cycle (int frames);
void one_cycle (int frames);
public:
static PortInfo port_info [];
d_sample * ports [2];
d_sample adding_gain;
void init (double _fs)
{ }
void init() {}
void activate()
{
gain = .5;

89
plugins/ladspa_effect/caps/basics.h
View File

@@ -46,19 +46,11 @@
#include <assert.h>
#include <stdio.h>
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#ifdef HAVE_LADSPA_H
#include <ladspa.h>
#else
#include "ladspa-1.1.h"
#endif
#include "ladspa.h"
#define BOUNDED (LADSPA_HINT_BOUNDED_BELOW | LADSPA_HINT_BOUNDED_ABOVE)
#define INTEGER LADSPA_HINT_INTEGER
#define FS LADSPA_HINT_SAMPLE_RATE
/* #define FS LADSPA_HINT_SAMPLE_RATE *//* deprecated */
#define LOG LADSPA_HINT_LOGARITHMIC
#define TOGGLE LADSPA_HINT_TOGGLED
@@ -84,16 +76,6 @@
#define MIN_GAIN .000001 /* -120 dB */
#define NOISE_FLOOR .00000000000005 /* -266 dB */
#ifdef BUILD_WIN32
typedef int8_t int8;
typedef uint8_t uint8;
typedef int16_t int16;
typedef uint16_t uint16;
typedef int32_t int32;
typedef uint32_t uint32;
typedef int64_t int64;
typedef uint64_t uint64;
#else
typedef __int8_t int8;
typedef __uint8_t uint8;
typedef __int16_t int16;
@@ -102,7 +84,6 @@ typedef __int32_t int32;
typedef __uint32_t uint32;
typedef __int64_t int64;
typedef __uint64_t uint64;
#endif
typedef struct {
char * name;
@@ -144,11 +125,73 @@ X max (X x, Y y)
}
#endif /* ! max */
template <class T>
T clamp (T value, T lower, T upper)
{
if (value < lower) return lower;
if (value > upper) return upper;
return value;
}
static inline float
frandom()
{
return (float) rand() / (float) RAND_MAX;
return (float) random() / (float) RAND_MAX;
}
/* for testing only. */
inline bool
is_denormal (float & f)
{
int32 i = *((int32 *) &f);
return ((i & 0x7f800000) == 0);
}
/* not sure if this double version is correct, actually ... */
inline bool
is_denormal (double & f)
{
int64 i = *((int64 *) &f);
return ((i & 0x7fe0000000000000ll) == 0);
}
#ifdef __i386__
#define TRAP asm ("int $3;")
#else
#define TRAP
#endif
/* //////////////////////////////////////////////////////////////////////// */
#define CAPS "C* "
class Plugin {
public:
double fs; /* sample rate */
double adding_gain; /* for run_adding() */
int first_run; /* 1st block after activate(), do no parameter smoothing */
d_sample normal; /* renormal constant */
d_sample ** ports;
LADSPA_PortRangeHint * ranges; /* for getport() below */
public:
/* get port value, mapping inf or nan to 0 */
inline d_sample getport_unclamped (int i)
{
d_sample v = *ports[i];
return (isinf (v) || isnan(v)) ? 0 : v;
}
/* get port value and clamp to port range */
inline d_sample getport (int i)
{
LADSPA_PortRangeHint & r = ranges[i];
d_sample v = getport_unclamped (i);
return clamp (v, r.LowerBound, r.UpperBound);
}
};
#endif /* _BASICS_H_ */

19
plugins/ladspa_effect/caps/dsp/BiQuad.h
View File

@@ -48,13 +48,11 @@ class BiQuad
reset();
}
BiQuad (d_sample * _a, d_sample * _b)
void copy (BiQuad & bq)
{
for (int i = 0; i < 3; ++i)
a[i] = _a[i],
b[i] = _b[i];
reset();
a[i] = bq.a[i],
b[i] = bq.b[i];
}
void reset()
@@ -65,6 +63,14 @@ class BiQuad
y[0] = y[1] = 0.;
}
/* denormal zapping */
void flush_0()
{
for (int i = 0; i < 2; ++i)
if (is_denormal (y[i]))
y[i] = 0;
}
inline d_sample process (d_sample s)
{
register int z = h;
@@ -86,7 +92,8 @@ class BiQuad
return r;
}
/* additional methods for using the biquad to upsample */
/* additional methods for using the biquad to filter an
* upsampled signal with 0 padding */
inline d_sample process_0_1()
{
register int z = h;

128
plugins/ladspa_effect/caps/dsp/Eq.h
View File

@@ -1,12 +1,12 @@
/*
Eq.h
Copyright 2004 Tim Goetze <tim@quitte.de>
Copyright 2004-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
equalizer circuit using recursive filtering.
based on a motorola paper implementing a similar circuit on a DSP56001.
Equalizer circuit using recursive filtering.
Based on a motorola paper implementing a similar circuit on a DSP56001.
*/
/*
@@ -31,86 +31,64 @@
namespace DSP {
/* a single bandpass as used by the Eq, expressed as a biquad. like all
* band-pass filters i know, the filter works with a FIR coefficient of 0
/* A single bandpass as used by the Eq, expressed as a biquad. Like all
* band-pass filters I know of, the filter works with a FIR coefficient of 0
* for x[-1], so a generic biquad isn't the optimum implementation.
*
* This routine isn't used anywhere, just here for testing purposes.
*/
class BP
template <class T>
void
_BP (double fc, double Q, T * ca, T * cb)
{
public:
template <class T>
BP (double fc, double Q, T * ca, T * cb)
{
double theta = 2 * fc * M_PI;
double theta = 2 * fc * M_PI;
double
b = (Q - theta * .5) / (2 * Q + theta),
a = (.5 - b) / 2,
c = (.5 + b) * cos (theta);
double
b = (Q - theta * .5) / (2 * Q + theta),
a = (.5 - b) / 2,
c = (.5 + b) * cos (theta);
ca[0] = 2 * a;
ca[1] = 0;
ca[2] = -2 * a;
ca[0] = 2 * a;
ca[1] = 0;
ca[2] = -2 * a;
cb[0] = 0;
cb[1] = 2 * c;
cb[2] = -2 * b;
}
};
cb[0] = 0;
cb[1] = 2 * c;
cb[2] = -2 * b;
}
/* BANDS must be a multiple of 4 to enable the use of SSE instructions.
* however, the current SSE-enabled process() method fails to compile if
* -funroll-loops is passed to gcc.
*/
template <int USE_BANDS, int BANDS>
template <int Bands, class eq_sample = float>
class Eq
{
public:
/* over-size state buffer to allow alignment */
float state [BANDS * 8 + 4 + 4];
/* recursion coefficients, 3 per band */
float * a, * b, * c;
eq_sample __attribute__ ((aligned)) a[Bands], b[Bands], c[Bands];
/* past outputs, 2 per band */
float * y;
eq_sample __attribute__ ((aligned)) y[2][Bands];
/* current gain and recursion factor, each 1 per band = 2 */
float * gain, * gf;
/* aligned storage for output summation */
float * temp;
/* aligned storage for constants */
float * two;
eq_sample __attribute__ ((aligned)) gain[Bands], gf[Bands];
/* input history */
float x[2];
eq_sample x[2];
/* history index */
int h;
eq_sample normal;
Eq()
{
/* take care of 128-bit alignment */
long s = (long) (char *) state;
s &= 0xF;
if (s)
s = 16 - s;
/* assign coefficients */
a = (float *) (((char *) state) + s);
b = a + BANDS;
c = a + 2 * BANDS;
/* output history (input is common to all bands) */
y = a + 3 * BANDS;
gain = a + 5 * BANDS;
gf = a + 6 * BANDS;
temp = a + 7 * BANDS;
two = temp + 4;
two[0] = two[1] = two[2] = two[3] = 2;
h = 0;
normal = NOISE_FLOOR;
}
void reset()
{
for (int i = 0; i < 2 * BANDS; ++i)
y[i] = 0;
{
for (int z = 0; z < 2; ++z)
{
// work-around for buggy optimizer in GCC 4.3
for (int i = 0; i < Bands-1; ++i)
y[z][i] = 0;
y[z][Bands-1] = 0;
}
for (int i = 0; i < 2; ++i)
x[i] = 0;
@@ -121,9 +99,10 @@ class Eq
double f = 31.25;
int i = 0;
for (i = 0; i < USE_BANDS && f < fs / 2; ++i, f *= 2)
for (i = 0; i < Bands && f < fs / 2; ++i, f *= 2)
init_band (i, 2 * f * M_PI / fs, Q);
for ( ; i < BANDS; ++i)
/* just in case, zero the remaining coefficients */
for ( ; i < Bands; ++i)
zero_band (i);
reset();
@@ -134,6 +113,7 @@ class Eq
b[i] = (Q - theta * .5) / (2 * Q + theta);
a[i] = (.5 - b[i]) / 2;
c[i] = (.5 + b[i]) * cos (theta);
/* fprintf (stderr, "%02d %f %f %f\n", i, a[i], b[i], c[i]); */
gain[i] = 1;
gf[i] = 1;
}
@@ -146,28 +126,36 @@ class Eq
/* per-band recursion:
* y = 2 * (a * (x - x[-2]) + c * y[-1] - b * y[-2])
*/
d_sample process (d_sample s)
eq_sample process (eq_sample s)
{
int z1 = h, z2 = h ^ 1;
float * y1 = y + z1 * BANDS;
float * y2 = y + z2 * BANDS;
eq_sample * y1 = y[z1];
eq_sample * y2 = y[z2];
d_sample x_x2 = s - x[z2];
d_sample r = 0;
eq_sample x_x2 = s - x[z2];
eq_sample r = 0;
for (int i = 0; i < USE_BANDS; ++i)
for (int i = 0; i < Bands; ++i)
{
y2[i] = 2 * (a[i] * x_x2 + c[i] * y1[i] - b[i] * y2[i]);
y2[i] = normal + 2 * (a[i] * x_x2 + c[i] * y1[i] - b[i] * y2[i]);
r += gain[i] * y2[i];
gain[i] *= gf[i];
}
x[z2] = s;
h = z2;
return r;
}
/* zap denormals in history */
void flush_0()
{
for (int i = 0; i < Bands; ++i)
if (is_denormal (y[0][i]))
y[0][i] = 0;
}
};
} /* namespace DSP */

12
plugins/ladspa_effect/caps/dsp/FPTruncateMode.h
View File

@@ -1,16 +1,19 @@
/* Copyright 2001-4 tim goetze <tim@quitte.de> -- see 'COPYING'. */
/* class that sets the FP rounding mode to 'truncate' in the constructor
/* Sets the FP rounding mode to 'truncate' in the constructor
* and loads the previous FP conrol word in the destructor.
*
* i386 implementation only, on other architectures this is a no-op.
* By directly using the machine instruction to convert float to int
* we avoid the performance hit that loading the control word twice for
* every (int) cast causes on i386.
*
* On other architectures this is a no-op.
*/
#ifndef _DSP_FP_TRUNCATE_MODE_H_
#define _DSP_FP_TRUNCATE_MODE_H_
#ifdef __i386__
#define fstcw(i) \
__asm__ __volatile__ ("fstcw %0" : "=m" (i))
@@ -20,15 +23,12 @@
/* gcc chokes on __volatile__ sometimes. */
#define fistp(f,i) \
__asm__ ("fistpl %0" : "=m" (i) : "t" (f) : "st")
#else /* ! __i386__ */
#define fstcw(i)
#define fldcw(i)
#define fistp(f,i) \
i = (int) f
#endif
namespace DSP {

102
plugins/ladspa_effect/caps/dsp/LatFilt.h Normal file
View File

@@ -0,0 +1,102 @@
/*
LatFilt.h
Copyright 2006 David Yeh <dtyeh@ccrma.stanford.edu>
Lattice digital filter.
Assumes order of b = order of a.
Assumes a0 = 1.
*/
/*
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
#ifndef _DSP_LatFilt_H_
#define _DSP_LatFilt_H_
namespace DSP {
// ORDER is the highest power of s in the transfer function
template <int ORDER>
class LatFilt
{
public:
double vcoef[ORDER+1];
double kcoef[ORDER];
double state[ORDER];
double y;
// fade factors
double vf[ORDER+1];
double kf[ORDER];
void reset()
{
for (int i = 0; i < ORDER; i++) {
state[i] = 0; // zero state
vf[i] = 1; // reset fade factor
kf[i] = 1;
}
vf[ORDER] = 1;
y = 0;
}
void init (double fs)
{
reset();
clearcoefs();
}
void clearcoefs() {
for (int i=0; i< ORDER; i++) {
vcoef[i] = 0;
kcoef[i] = 0;
}
vcoef[ORDER] = 0;
}
d_sample process (d_sample s) {
double tmp;
int i = ORDER-1;
tmp = -kcoef[i]*state[i] + s;
y = vcoef[i+1]*(state[i] + kcoef[i]*tmp);
for (i = ORDER-2; i >= 0; i--) {
tmp = -kcoef[i]*state[i] + tmp;
state[i+1] = kcoef[i]*tmp + state[i];
y = y + vcoef[i+1]*state[i+1];
}
state[0] = tmp;
y = y + vcoef[0]*tmp;
return (d_sample) y;
}
inline void set_vi(double coef, int i) {
vcoef[i] = coef;
}
inline void set_ki(double coef, int i) {
kcoef[i] = coef;
}
};
} /* namespace DSP */
#endif /* _DSP_LatFilt_H_ */

19
plugins/ladspa_effect/caps/dsp/OnePole.h
View File

@@ -67,25 +67,29 @@ class OnePoleLP
a0 *= d;
b1 = 1. - a0;
}
/* clear denormal numbers in history */
void flush_0()
{
if (is_denormal (y1))
y1 = 0;
}
};
class OnePoleHP
{
public:
d_sample a0, a1, b1, x1, y1;
double fc;
OnePoleHP (double d = 1.)
{
fc = 0;
set (d);
x1 = y1 = 0.;
}
void set_f (double f)
{
fc = f;
set (exp (-2 * M_PI * fc));
set (exp (-2 * M_PI * f));
}
inline void set (double d)
@@ -106,6 +110,13 @@ class OnePoleHP
{
x1 = y1 = 0;
}
/* clear denormal numbers in history */
void flush_0()
{
if (is_denormal (y1))
y1 = 0;
}
};
} /* namespace DSP */

18
plugins/ladspa_effect/caps/dsp/RMS.h
View File

@@ -49,14 +49,24 @@ class RMS
memset (buffer, 0, sizeof (buffer));
}
/* needs the squared sample value to be passed in */
d_sample process (d_sample x)
/* caution: pass in the *squared* sample value */
void store (d_sample x)
{
sum -= buffer[write];
sum += x;
sum += (buffer[write] = x);
write = (write + 1) & 63;
}
return sqrt (sum / 64);
d_sample process (d_sample x)
{
store (x);
return rms();
}
d_sample rms()
{
/* fabs it before sqrt, just in case ... */
return sqrt (fabs (sum) / 64);
}
};

80
plugins/ladspa_effect/caps/dsp/TDFII.h Normal file
View File

@@ -0,0 +1,80 @@
/*
TDFII.h
Copyright 2006-7
David Yeh <dtyeh@ccrma.stanford.edu> (implementation)
Tim Goetze <tim@quitte.de> (cosmetics)
transposed Direct Form II digital filter.
Assumes order of b = order of a.
Assumes a0 = 1.
*/
/*
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
#ifndef _DSP_TDFII_H_
#define _DSP_TDFII_H_
namespace DSP {
// ORDER is the highest power of s in the transfer function
template <int Order>
class TDFII
{
public:
double a[Order + 1];
double b[Order + 1];
double h[Order + 1];
void reset()
{
for (int i = 0; i <= Order; ++i)
h[i] = 0; // zero state
}
void init (double fs)
{
reset();
clear();
}
void clear()
{
for (int i=0; i<= Order; i++)
a[i] = b[i] = 1;
}
/* per-band recursion:
* y = 2 * (a * (x - x[-2]) + c * y[-1] - b * y[-2])
*/
d_sample process (d_sample s)
{
double y = h[0] + b[0] * s;
for (int i = 1; i < Order; ++i)
h[i - 1] = h[i] + b[i] * s - a[i] * y;
h[Order - 1] = b[Order] * s - a[Order] * y;
return (d_sample) y;
}
};
} /* namespace DSP */
#endif /* _DSP_TDFII_H_ */

268
plugins/ladspa_effect/caps/dsp/ToneStack.h Normal file
View File

@@ -0,0 +1,268 @@
/*
ToneStack.h
Copyright 2006-7
David Yeh <dtyeh@ccrma.stanford.edu> (implementation)
Tim Goetze <tim@quitte.de> (cosmetics)
Tone Stack emulation.
*/
/*
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
#ifndef _DSP_TONESTACK_H_
#define _DSP_TONESTACK_H_
#include "util.h"
#include "windows.h"
#include "TDFII.h"
#include "LatFilt.h"
#include "tonestack/tables.h"
namespace DSP {
typedef struct {
double R1, R2, R3, R4;
double C1, C2, C3;
} TSParameters;
#define TS_N_PRESETS (sizeof (DSP::ToneStack::presets) / \
sizeof (DSP::TSParameters))
class ToneStack
{
private:
enum { Order = 3 };
double c; // BT coef
double b1t, b1m, b1l, b1d,
b2t, b2m2, b2m, b2l, b2lm, b2d,
b3lm, b3m2, b3m, b3t, b3tm, b3tl,
a0, a1d, a1m, a1l, a2m, a2lm, a2m2, a2l, a2d,
a3lm, a3m2, a3m, a3l, a3d; // intermediate calculations
struct {
double b1;
double b2;
double b3;
double a1;
double a2;
double a3;
} acoef; // analog coefficients
// digital coefficients
double dcoef_a[Order + 1];
double dcoef_b[Order + 1];
double af[Order + 1];
double bf[Order + 1];
double fs;
TDFII<Order> filter;
public:
int model;
static TSParameters presets[]; /* in ../ToneStack.cc */
static int n_presets;
ToneStack()
{
setparams (presets[0]);
}
void init (double _fs)
{
c = 2 * _fs;
}
void activate (d_sample ** ports)
{
filter.reset();
}
/* pass in pointer to ports and relative index of first eq band control */
void start_cycle (d_sample ** ports, int bassindex = 1)
{
int m = clamp<int> ((int) *ports[0], 0, n_presets - 1);
if (m != model)
{
model = m;
setparams (presets[model]);
filter.reset();
}
updatecoefs (ports + bassindex);
}
void setparams (TSParameters & p)
{
double R1 = p.R1, R2 = p.R2, R3 = p.R3, R4 = p.R4;
double C1 = p.C1, C2 = p.C2, C3 = p.C3;
b1t = C1*R1;
b1m = C3*R3;
b1l = C1*R2 + C2*R2;
b1d = C1*R3 + C2*R3;
b2t = C1*C2*R1*R4 + C1*C3*R1*R4;
b2m2 = -(C1*C3*R3*R3 + C2*C3*R3*R3);
b2m = C1*C3*R1*R3 + C1*C3*R3*R3 + C2*C3*R3*R3;
b2l = C1*C2*R1*R2 + C1*C2*R2*R4 + C1*C3*R2*R4;
b2lm = C1*C3*R2*R3 + C2*C3*R2*R3;
b2d = C1*C2*R1*R3 + C1*C2*R3*R4 + C1*C3*R3*R4;
b3lm = C1*C2*C3*R1*R2*R3 + C1*C2*C3*R2*R3*R4;
b3m2 = -(C1*C2*C3*R1*R3*R3 + C1*C2*C3*R3*R3*R4);
b3m = C1*C2*C3*R1*R3*R3 + C1*C2*C3*R3*R3*R4;
b3t = C1*C2*C3*R1*R3*R4;
b3tm = -b3t;
b3tl = C1*C2*C3*R1*R2*R4;
a0 = 1;
a1d = C1*R1 + C1*R3 + C2*R3 + C2*R4 + C3*R4;
a1m = C3*R3;
a1l = C1*R2 + C2*R2;
a2m = C1*C3*R1*R3 - C2*C3*R3*R4 + C1*C3*R3*R3 + C2*C3*R3*R3;
a2lm = C1*C3*R2*R3 + C2*C3*R2*R3;
a2m2 = -(C1*C3*R3*R3 + C2*C3*R3*R3);
a2l = C1*C2*R2*R4 + C1*C2*R1*R2 + C1*C3*R2*R4 + C2*C3*R2*R4;
a2d = C1*C2*R1*R4 + C1*C3*R1*R4 + C1*C2*R3*R4
+ C1*C2*R1*R3 + C1*C3*R3*R4 + C2*C3*R3*R4;
a3lm = C1*C2*C3*R1*R2*R3 + C1*C2*C3*R2*R3*R4;
a3m2 = -(C1*C2*C3*R1*R3*R3 + C1*C2*C3*R3*R3*R4);
a3m = C1*C2*C3*R3*R3*R4 + C1*C2*C3*R1*R3*R3 - C1*C2*C3*R1*R3*R4;
a3l = C1*C2*C3*R1*R2*R4;
a3d = C1*C2*C3*R1*R3*R4;
filter.reset();
}
inline void updatecoefs (d_sample ** ports)
{
/* range checks on input */
double b = clamp<double> (*ports[0], 0, 1);
double m = clamp<double> (*ports[1], 0, 1);
double t = clamp<double> (*ports[2], 0, 1);
m = (m - 1) * 3.5;
m = pow (10, m);
acoef.a1 = a1d + m*a1m + b*a1l;
acoef.a2 = m*a2m + b*m*a2lm + m*m*a2m2 + b*a2l + a2d;
acoef.a3 = b*m*a3lm + m*m*a3m2 + m*a3m + b*a3l + a3d;
dcoef_a[0] = -1 - acoef.a1*c - acoef.a2*c*c - acoef.a3*c*c*c; // sets scale
dcoef_a[1] = -3 - acoef.a1*c + acoef.a2*c*c + 3*acoef.a3*c*c*c;
dcoef_a[2] = -3 + acoef.a1*c + acoef.a2*c*c - 3*acoef.a3*c*c*c;
dcoef_a[3] = -1 + acoef.a1*c - acoef.a2*c*c + acoef.a3*c*c*c;
acoef.b1 = t*b1t + m*b1m + b*b1l + b1d;
acoef.b2 = t*b2t + m*m*b2m2 + m*b2m + b*b2l + b*m*b2lm + b2d;
acoef.b3 = b*m*b3lm + m*m*b3m2 + m*b3m + t*b3t + t*m*b3tm + t*b*b3tl;
dcoef_b[0] = - acoef.b1*c - acoef.b2*c*c - acoef.b3*c*c*c;
dcoef_b[1] = - acoef.b1*c + acoef.b2*c*c + 3*acoef.b3*c*c*c;
dcoef_b[2] = acoef.b1*c + acoef.b2*c*c - 3*acoef.b3*c*c*c;
dcoef_b[3] = acoef.b1*c - acoef.b2*c*c + acoef.b3*c*c*c;
for (int i=1; i<=3; ++i)
filter.a[i] = dcoef_a[i] / dcoef_a[0];
for (int i=0; i<=3; ++i)
filter.b[i] = dcoef_b[i] / dcoef_a[0];
}
// actualy do the DFII filtering, one sample at a time
inline d_sample process (d_sample x)
{
return filter.process (x);
}
};
/* /////////////////////////////////////////////////////////////////////// */
/*
hardcode this, known size memory blocks
extern double* KS; // 25 x 25 x 3
extern double* VS; // 25 x 25 x 25 x 4
extern double KS[NSTEPS][NSTEPS][TSORDER]; //[bass][mid][coefs]
extern double VS[NSTEPS][NSTEPS][NSTEPS][TSORDER+1]; //[bass][mid][treb][coefs]
*/
class ToneStackLT
{
private:
enum { Order = 3, Steps = 25 };
// digital coefficients
double *kcoef;
double *vcoef;
double af [Order + 1];
double bf [Order + 1];
double fs;
LatFilt<Order> filter;
public:
ToneStackLT()
{
setparams (250000, 1000000, 25000, 56000, 0.25e-9, 20e-9, 20e-9);
}
void init (double _fs)
{ }
void activate (d_sample ** ports)
{
filter.reset();
}
void setparams
(double R1, double R2, double R3, double R4,
double C1, double C2, double C3) {
int blah[4];
int *bp;
bp = blah;
}
void updatecoefs (d_sample ** ports)
{
double b = min (Steps - 1, max (*ports[0] * (Steps - 1), 0));
double m = min (Steps - 1, max (*ports[1] * (Steps - 1), 0));
double t = min (Steps - 1, max (*ports[2] * (Steps - 1), 0));
int bi = (int) b;
int mi = (int) m;
int ti = (int) t;
kcoef = DSP::ToneStackKS + (mi * Steps + bi) * Order;
vcoef = DSP::ToneStackVS + ((mi * Steps + bi) * Steps + ti) * (Order + 1);
for (int i = 0; i < Order; ++i)
filter.set_ki (kcoef[i], i);
for (int i = 0; i < Order + 1; ++i)
filter.set_vi (vcoef[i], i);
}
// actualy do the DFII filtering, one sample at a time
inline d_sample process (d_sample x)
{
return filter.process (x);
}
};
} /* namespace DSP */
#endif /* _DSP_TONESTACK_H_ */

44
plugins/ladspa_effect/caps/dsp/TwelveAX7.h
View File

@@ -31,6 +31,8 @@ namespace DSP {
#include "r12ax7.h"
typedef d_sample tube_sample;
/* this is the original tube model from caps < 0.1.9 or preamp.so, put
* back into use in 0.1.11; the replacement (below) is too strong in
* odd-order harmonics at the expense of even-order. it has to sound
@@ -39,14 +41,14 @@ namespace DSP {
class TwelveAX7
{
public:
d_sample b, c, d;
tube_sample b, c, d;
struct {
d_sample threshold, value;
tube_sample threshold, value;
} clip[2];
/* amplitude at which clipping starts */
d_sample scale;
tube_sample scale;
public:
TwelveAX7()
@@ -61,12 +63,12 @@ class TwelveAX7
scale = min (fabs (clip[0].threshold), fabs (clip[1].threshold));
}
inline d_sample transfer (d_sample a)
inline tube_sample transfer (tube_sample a)
{
return a * (b + a * (c + a * d));
}
inline d_sample transfer_clip (d_sample a)
inline tube_sample transfer_clip (tube_sample a)
{
if (a <= clip[0].threshold)
return clip[0].value;
@@ -103,13 +105,13 @@ class TwelveAX7
class TwelveAX7_2
{
public:
d_sample b, c, d;
tube_sample b, c, d;
struct {
d_sample threshold, value;
tube_sample threshold, value;
} clip[2];
d_sample scale;
tube_sample scale;
public:
TwelveAX7_2()
@@ -128,12 +130,12 @@ class TwelveAX7_2
scale = min (fabs (clip[0].threshold), fabs (clip[1].threshold));
}
inline d_sample transfer (d_sample a)
inline tube_sample transfer (tube_sample a)
{
return a * (b + a * (c + a * d));
}
inline d_sample transfer_clip (d_sample a)
inline tube_sample transfer_clip (tube_sample a)
{
if (a <= clip[0].threshold)
return clip[0].value;
@@ -149,22 +151,20 @@ class TwelveAX7_2
class TwelveAX7_3
{
public:
d_sample b, c, d;
tube_sample b, c, d;
struct {
d_sample threshold, value;
tube_sample threshold, value;
} clip[2];
d_sample scale;
tube_sample scale;
public:
TwelveAX7_3()
{
static double x[2] =
{
/* adjust for a slightly earlier clipping threshold in the
* lower lobe */
-0.96 * (double) r12AX7::Zero /
(double) r12AX7::Zero /
((double) r12AX7::Samples - (double) r12AX7::Zero),
1
};
@@ -176,7 +176,7 @@ class TwelveAX7_3
scale = min (fabs (clip[0].threshold), fabs (clip[1].threshold));
}
inline d_sample transfer (d_sample a)
inline tube_sample transfer (tube_sample a)
{
a = r12AX7::Zero + a * (r12AX7::Samples - r12AX7::Zero);
if (a <= 0)
@@ -191,7 +191,7 @@ class TwelveAX7_3
return (r12AX7::v2v [i] * (1.f - a) + r12AX7::v2v [i + 1] * a);
}
inline d_sample transfer_clip (d_sample a)
inline tube_sample transfer_clip (tube_sample a)
{
return transfer (a);
}
@@ -202,11 +202,11 @@ class NoTwelveAX7
{
public:
struct {
d_sample threshold, value;
tube_sample threshold, value;
} clip[2];
/* amplitude at which clipping starts */
d_sample scale;
tube_sample scale;
public:
NoTwelveAX7()
@@ -220,12 +220,12 @@ class NoTwelveAX7
scale = min (fabs (clip[0].threshold), fabs (clip[1].threshold));
}
inline d_sample transfer (d_sample a)
inline tube_sample transfer (tube_sample a)
{
return 0.5469181606780 * (pow (1 - a, 1.5) - 1);
}
inline d_sample transfer_clip (d_sample a)
inline tube_sample transfer_clip (tube_sample a)
{
if (a <= clip[0].threshold)
return clip[0].value;

663
plugins/ladspa_effect/caps/dsp/tonestack/ks_tab.h Normal file
View File

@@ -0,0 +1,663 @@
/*
ks_tab.h
Copyright 2006 David Yeh <dtyeh@ccrma.stanford.edu>
http://quitte.de/dsp/
Tone Stack emulation coefficient table for lattice filter.
*/
/*
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
#ifndef _KS_TAB_H_
#define _KS_TAB_H_
namespace DSP {
double ToneStackKS[] = {
-0.999655160907038,0.980127109825519,-0.614914419461634,
-0.999656568120978,0.980206907765841,-0.615188716843305,
-0.999658517676675,0.980317236486758,-0.615568128036899,
-0.999661208271173,0.98046907704722,-0.616090612857295,
-0.999664902023365,0.980676730894581,-0.616805748794314,
-0.999669936365576,0.980958270564762,-0.617776442867139,
-0.999676730565233,0.981335546377295,-0.619079214371838,
-0.999685778922586,0.981833264634174,-0.620801399623313,
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-0.999368702795487,0.938696629074492,-0.485653574382402,
-0.999431161107827,0.948513477384418,-0.516600823762431,
-0.999499889019962,0.957794017950755,-0.547100205403036,
-0.999571867439423,0.966117436282565,-0.575553712340007,
-0.999643400523047,0.973236877263828,-0.600771919790487,
-0.999710829097846,0.979083725305741,-0.622127004150167,
-0.999771250966691,0.983727246264749,-0.639523867161234,
-0.999822958889432,0.987318112980124,-0.653254531680796,
-0.999865479385908,0.990038301447359,-0.663823305665926,
-0.999899304831475,0.992067025349649,-0.67180239279328,
-0.999925506987289,0.993562576625782,-0.677738773058922,
-0.999945387183711,0.994655694348136,-0.682107445763671,
-0.999960233952359,0.995449700414856,-0.685296658764391,
-0.999971190912995,0.996023838521735,-0.687611196422924,
-0.999979206471127,0.996437638512654,-0.689283790136547,
-0.999985032608414,0.996735177154657,-0.690488758695185,
-0.999989247549769,0.996948758152623,-0.691354911475921,
-0.999992286527841,0.997101886480435,-0.691976522006297,
-0.999042007259475,0.846839476355646,-0.247562606935731,
-0.999059310027056,0.853850930614133,-0.262993110426348,
-0.99908215215834,0.862634865772264,-0.282874416523585,
-0.999111791350179,0.873267701688246,-0.307799227855353,
-0.999149460274721,0.885615899778428,-0.337997269244,
-0.999196176211498,0.89928385557361,-0.373105577817442,
-0.999252479946996,0.913630787322957,-0.412017325303364,
-0.999318122840983,0.927874730147937,-0.452924609459096,
-0.9993917698708,0.941254666200446,-0.493608708966554,
-0.999470858355026,0.953184064404172,-0.531898307960971,
-0.999551781187113,0.963334777927113,-0.566107560700536,
-0.999630458843132,0.971634043939743,-0.595279512665676,
-0.999703140993037,0.978200975484896,-0.619183392925497,
-0.999767110732338,0.983264056319206,-0.638138631171118,
-0.999821016253149,0.987090375099302,-0.652782585207141,
-0.999864780149848,0.989938753401259,-0.663869411221801,
-0.999899238810493,0.992035545799555,-0.672135353966679,
-0.999925718023562,0.993566460874601,-0.678227911673286,
-0.999945685530231,0.99467756984301,-0.682680702983936,
-0.999960528925675,0.995480521419424,-0.685914973874208,
-0.999971446026693,0.996058980854992,-0.688253627504821,
-0.999979412416931,0.996474783268042,-0.689939175953683,
-0.999985192241392,0.996773187954865,-0.69115115989633,
-0.999989368141114,0.996987095415323,-0.692021159457422,
-0.999992376091346,0.997140306222664,-0.692644915228308,
-0.998865086928174,0.745924580066976,-0.0604141480564288,
-0.998895487231205,0.764873649745473,-0.0910755754690588,
-0.998933409668284,0.787060755596958,-0.129313938502959,
-0.998979684019541,0.81184986561543,-0.175344827159791,
-0.99903493644843,0.83813626350615,-0.228455101442363,
-0.999099521125361,0.864509230253996,-0.286815983154912,
-0.999173362695985,0.889538052977561,-0.347607059890004,
-0.999255664139382,0.912066866492635,-0.407502232575221,
-0.999344565973438,0.931397810612656,-0.463365245732482,
-0.99943698944938,0.94731384445106,-0.512864721985471,
-0.999528900858466,0.959977803907794,-0.554772924977228,
-0.999616021217074,0.969782650772129,-0.588907534659359,
-0.999694709613298,0.977214767105395,-0.615844974558559,
-0.999762624036077,0.982758778214894,-0.636577066889482,
-0.999818918115288,0.986845473409128,-0.652227863178924,
-0.999864010307122,0.989831859802084,-0.663871226969659,
-0.999899137899114,0.992000508059639,-0.672439368989748,
-0.999925909089307,0.993568232917785,-0.6786940662434,
-0.999945970648199,0.994697887686373,-0.68323323728191,
-0.999960814301116,0.995510003466929,-0.686513398356392,
-0.99997169391947,0.996092876111869,-0.688876459454257,
-0.999979612900072,0.996510723831304,-0.690575054289614,
-0.999985347769733,0.996810017409768,-0.69179407428053,
-0.999989485678529,0.997024265113336,-0.69266791809088,
-0.999992463403904,0.997177567482597,-0.693293810560046,
-0.998168670041896,0.234202002930654,0.397757295018613,
-0.998398779709349,0.385186319412462,0.309863134853804,
-0.998576095338357,0.517988723723858,0.208214440973434,
-0.998719357216826,0.629703569518859,0.096222948116353,
-0.998842476527936,0.720203330951345,-0.0207601551784617,
-0.998955556213729,0.791300079843907,-0.136348134920346,
-0.999065378019792,0.845813571958666,-0.24445016793907,
-0.999175567184225,0.88683389735994,-0.340477179725281,
-0.999286737933873,0.917262868364474,-0.421953754912466,
-0.999396947303236,0.939594279052761,-0.488436998558289,
-0.999502626750312,0.955853286353504,-0.540978607500742,
-0.999599838919708,0.967622793392235,-0.581462483326913,
-0.999685441175006,0.976107385917766,-0.612050478470221,
-0.999757759055763,0.982206390023536,-0.634821102862626,
-0.999816654953809,0.986582054280449,-0.651585654275972,
-0.999863167558551,0.989717294446952,-0.663828002921204,
-0.999899001887773,0.991961866430869,-0.672714826358979,
-0.999926080483689,0.993567921696033,-0.679137945776486,
-0.999946242902525,0.994716695840987,-0.683765847313538,
-0.999961090399614,0.995538198798929,-0.687092756503707,
-0.999971934849436,0.996125577031366,-0.689480524558436,
-0.999979808120392,0.996545512563991,-0.691192260465585,
-0.999985499344625,0.996845717383694,-0.692418338685915,
-0.999989600274749,0.997060318675938,-0.693296025163123,
-0.999992548548667,0.997213721346586,-0.693924046451602
};
} /* namespace DSP */
#endif /* _KS_TAB_H_ */

38
plugins/ladspa_effect/caps/dsp/tonestack/tables.h Normal file
View File

@@ -0,0 +1,38 @@
/*
ks_tab.h
Copyright 2006 David Yeh <dtyeh@ccrma.stanford.edu>
http://quitte.de/dsp/
Tone Stack emulation coefficient table for lattice filter.
*/
/*
This program is free software; you can redistribute it and/or
modify it under the terms of the GNU General Public License
as published by the Free Software Foundation; either version 2
of the License, or (at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
02111-1307, USA or point your web browser to http://www.gnu.org.
*/
#ifndef _TS_TABLES_H_
#define _TS_TABLES_H_
namespace DSP {
extern double ToneStackKS[];
extern double ToneStackVS[];
} /* namespace DSP */
#endif /* _TS_TABLES_H_ */

15664
plugins/ladspa_effect/caps/dsp/tonestack/vs_tab.h Normal file
View File

File diff suppressed because it is too large Load Diff

2
plugins/ladspa_effect/caps/dsp/util.h
View File

@@ -5,7 +5,7 @@
http://quitte.de/dsp/
common math utility functions.
Common math utility functions.
*/
/*

22
plugins/ladspa_effect/caps/interface.cc
View File

@@ -1,6 +1,10 @@
/*
interface.cc
Copyright 2004-7 Tim Goetze <tim@quitte.de>
http://quitte.de/dsp/
LADSPA descriptor factory, host interface.
*/
@@ -43,10 +47,11 @@
#include "HRTF.h"
#include "Pan.h"
#include "Scape.h"
#include "ToneStack.h"
#include "Descriptor.h"
#define N 33
#define N 38
static DescriptorStub * descriptors [N];
static inline void
@@ -55,24 +60,29 @@ seed()
static struct timeval tv;
gettimeofday (&tv, 0);
srand (tv.tv_sec ^ tv.tv_usec);
srandom (tv.tv_sec ^ tv.tv_usec);
}
extern "C" {
__attribute__ ((constructor))
void _init()
{
DescriptorStub ** d = descriptors;
*d++ = new Descriptor<Eq>();
*d++ = new Descriptor<Eq2x2>();
*d++ = new Descriptor<Compress>();
*d++ = new Descriptor<Pan>();
*d++ = new Descriptor<PreampIII>();
*d++ = new Descriptor<PreampIV>();
*d++ = new Descriptor<ToneStack>();
*d++ = new Descriptor<ToneStackLT>();
*d++ = new Descriptor<AmpIII>();
*d++ = new Descriptor<AmpIV>();
*d++ = new Descriptor<AmpV>();
*d++ = new Descriptor<AmpVTS>();
*d++ = new Descriptor<CabinetI>();
*d++ = new Descriptor<CabinetII>();
*d++ = new Descriptor<Clip>();
@@ -85,6 +95,7 @@ void _init()
*d++ = new Descriptor<PhaserII>();
*d++ = new Descriptor<SweepVFI>();
*d++ = new Descriptor<SweepVFII>();
*d++ = new Descriptor<AutoWah>();
*d++ = new Descriptor<Scape>();
*d++ = new Descriptor<VCOs>();
@@ -103,9 +114,13 @@ void _init()
*d++ = new Descriptor<Dirac>();
*d++ = new Descriptor<HRTF>();
seed();
/* make sure N is correct */
assert (d - descriptors == N);
//seed();
}
__attribute__ ((destructor))
void _fini()
{
for (ulong i = 0; i < N; ++i)
@@ -119,7 +134,6 @@ ladspa_descriptor (unsigned long i)
{
if (i < N)
return descriptors[i];
return 0;
}

520
plugins/ladspa_effect/caps/waves/click.h Normal file
View File

@@ -0,0 +1,520 @@
float click [] = {
-0.013062, -0.013062, -0.012817, -0.011078, -0.031097, -0.266479,
-0.610718, -0.255005, 0.698975, 0.999969, 0.505188, -0.724579,
-1.000000, -0.493225, 0.532135, 0.789459, 0.131744, -0.270599,
-0.080780, -0.065796, 0.584381, 0.718567, -0.355316, -0.760559,
-0.649994, -0.039032, 0.442017, 0.700043, 0.434845, -0.556427,
-1.000000, -0.400391, 0.590729, 0.999969, 0.999969, 0.592896,
-0.248932, -0.662750, -0.074615, 0.424774, 0.274933, -0.416504,
-0.772736, -0.359833, -0.218140, -0.254456, -0.379547, -0.768280,
-0.707428, 0.108124, 0.755127, 0.680847, 0.429352, 0.126129,
-0.424316, -0.451813, 0.197510, 0.862762, 0.964508, 0.263824,
-0.383057, -0.753571, -0.848297, -0.491425, -0.042114, -0.222504,
-0.664490, -0.690155, -0.227905, 0.286682, 0.742981, 0.812225,
0.326691, 0.182770, 0.381195, 0.536194, 0.446930, 0.085754,
-0.650604, -1.000000, -1.000000, -0.774658, -0.152191, -0.112122,
-0.416290, -0.491547, -0.253235, 0.241760, 0.749847, 0.618042,
0.338440, 0.431335, 0.773712, 0.936890, 0.752563, 0.291595,
-0.206970, -0.552460, -0.447449, -0.152069, -0.315735, -0.891846,
-1.000000, -1.000000, -0.682281, 0.066925, 0.477264, 0.447144,
0.209167, 0.149994, 0.374176, 0.621185, 0.597046, 0.315826,
0.034210, -0.004089, 0.084534, 0.127075, 0.060608, -0.134949,
-0.275421, -0.347870, -0.505371, -0.618835, -0.443878, -0.217682,
-0.157898, -0.241058, -0.424103, -0.494690, -0.286438, 0.111542,
0.487793, 0.654907, 0.571869, 0.484650, 0.413208, 0.254089,
0.136414, 0.021118, -0.223724, -0.366364, -0.290771, -0.126251,
-0.080048, -0.175262, -0.290100, -0.281403, -0.071808, 0.078613,
0.216339, 0.275787, 0.223938, 0.178131, 0.162659, 0.110779,
0.104553, 0.158661, 0.208252, 0.271484, 0.256531, 0.153046,
0.068054, -0.009003, -0.116119, -0.155304, -0.099213, 0.066589,
0.293732, 0.392303, 0.313416, 0.196320, 0.123932, 0.145874,
0.279053, 0.291016, 0.081818, -0.092133, -0.090363, 0.030029,
0.099762, -0.021912, -0.133942, -0.025299, 0.257782, 0.466980,
0.406006, 0.150085, -0.007385, -0.027588, 0.048920, 0.128998,
0.058868, -0.066589, -0.146759, -0.168427, -0.119751, -0.087677,
-0.082581, -0.029694, -0.041138, -0.139984, -0.156036, -0.033325,
0.164612, 0.331726, 0.320435, 0.205811, 0.209717, 0.275909,
0.240479, 0.081238, -0.104828, -0.239624, -0.286652, -0.254120,
-0.304688, -0.428223, -0.479248, -0.534729, -0.453003, -0.144928,
-0.001709, 0.059448, 0.016937, -0.127991, -0.186432, -0.122986,
0.086090, 0.278351, 0.258026, 0.172882, -0.032043, -0.267242,
-0.289276, -0.293365, -0.288971, -0.256073, -0.212250, -0.214783,
-0.182709, -0.149170, -0.132599, -0.221649, -0.233063, 0.002472,
0.208008, 0.119659, -0.018250, -0.182861, -0.267731, -0.164001,
-0.052063, 0.009979, 0.058472, 0.147156, 0.111298, 0.017426,
-0.067780, -0.049835, 0.002167, -0.028259, 0.003418, 0.023743,
-0.056702, -0.074066, -0.070984, -0.075439, -0.056915, -0.189575,
-0.324524, -0.241882, -0.051514, 0.075623, 0.160095, 0.094238,
0.004425, 0.010193, 0.032898, 0.040222, 0.163605, 0.273682,
0.192261, -0.019989, -0.217682, -0.327728, -0.337372, -0.227386,
-0.043945, 0.107788, 0.244690, 0.226227, -0.034332, -0.252106,
-0.364136, -0.377136, -0.246277, -0.021973, 0.194763, 0.300110,
0.295532, 0.246246, 0.198425, 0.209991, 0.272064, 0.249451,
0.078094, -0.143616, -0.276459, -0.257782, -0.228638, -0.161224,
-0.006012, 0.169342, 0.267456, 0.246094, 0.232147, 0.290436,
0.280670, 0.220886, 0.162811, 0.187286, 0.178223, 0.036804,
-0.043304, -0.098694, -0.101990, 0.008698, 0.102844, 0.132050,
0.142578, 0.152924, 0.058960, -0.074615, -0.079041, 0.026123,
0.136322, 0.253510, 0.382782, 0.447357, 0.419800, 0.294952,
0.129059, 0.004028, -0.119781, -0.257538, -0.320526, -0.211609,
-0.054565, 0.012085, -0.016663, 0.006592, 0.144958, 0.193237,
0.072174, -0.018951, 0.038483, 0.085175, -0.023407, -0.089661,
-0.029999, -0.023956, -0.079498, -0.135223, -0.154205, -0.095276,
-0.031830, -0.105988, -0.252563, -0.284424, -0.153198, 0.017273,
0.073456, 0.072571, 0.175751, 0.222260, 0.105316, -0.053345,
-0.180115, -0.192932, -0.185211, -0.167511, -0.079529, -0.070007,
-0.151276, -0.215210, -0.206390, -0.133820, -0.099060, -0.082855,
0.047852, 0.108429, 0.014099, -0.049377, -0.050751, -0.081207,
-0.060944, -0.028564, -0.124420, -0.139526, -0.033508, 0.054108,
0.042664, -0.004669, -0.069611, -0.105774, -0.090881, -0.128601,
-0.090027, -0.012634, 0.004578, 0.046387, 0.085144, 0.055511,
0.036469, 0.001770, -0.019867, 0.016998, 0.031891, 0.089935,
0.066254, -0.069183, -0.020081, 0.046356, 0.005554, -0.023224,
-0.042847, -0.133057, -0.104004, 0.065674, 0.146484, 0.145782,
0.006897, -0.059509, 0.070923, 0.058319, -0.002655, 0.000641,
-0.036011, 0.049988, 0.076752, 0.016998, 0.063049, 0.105835,
0.041870, -0.092072, -0.154358, -0.081879, 0.089905, 0.090820,
0.054047, -0.110565, -0.285980, -0.219116, -0.103058, 0.083405,
0.192200, 0.124146, 0.080994, 0.118896, 0.231659, 0.259644,
0.175018, 0.110840, 0.005096, -0.156128, -0.202209, -0.167572,
-0.145050, -0.126068, -0.139221, -0.039856, 0.078674, 0.062866,
-0.069672, -0.143311, -0.049530, 0.082611, 0.142487, 0.039612,
-0.035736, -0.006287, 0.017181, 0.013580, -0.019043, -0.017303,
0.007904, -0.073944, -0.121704, -0.065369, -0.098083, -0.063568,
0.056885, 0.083984, 0.148987, 0.298431, 0.253815, 0.052643,
-0.139740, -0.217743, -0.073181, 0.097351, 0.124786, 0.091156,
0.056152, -0.007446, 0.048553, 0.129333, 0.021118, -0.003235,
0.093872, 0.080658, 0.025482, -0.037598, -0.055481, -0.078796,
-0.130676, -0.140930, -0.126770, -0.048004, 0.026093, 0.035767,
0.049957, 0.009247, -0.076599, -0.066101, 0.002747, 0.076813,
0.173492, 0.091949, -0.071960, -0.080902, -0.061157, -0.061066,
-0.073029, -0.070251, -0.008545, 0.035187, -0.036194, -0.086456,
-0.158691, -0.199036, -0.150177, -0.099030, -0.027802, -0.001465,
0.003662, 0.089020, 0.056122, -0.108551, -0.125610, -0.056610,
-0.053284, -0.128998, -0.166504, -0.124725, -0.095459, -0.088531,
-0.087433, -0.065399, -0.084045, -0.053284, -0.006317, -0.008698,
0.025879, -0.019714, -0.055511, -0.076874, -0.084442, -0.111450,
-0.097198, -0.040924, -0.047729, 0.025604, 0.077301, 0.027100,
-0.052612, -0.046265, 0.021271, 0.022186, -0.060669, -0.109283,
-0.022858, 0.126740, 0.199036, 0.255798, 0.210114, 0.167450,
0.034515, -0.140167, -0.106354, -0.056549, -0.016449, 0.092529,
0.082977, 0.035645, 0.066132, 0.113098, 0.123657, 0.150513,
0.123840, 0.125702, 0.146637, 0.131683, 0.114014, 0.019806,
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2715
plugins/ladspa_effect/caps/waves/money.h Normal file
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